- sr-users - kamailio.org

Sync Suspended Transaction
by Knserbrave 30 Sep '24

30 Sep '24

Is it possible to sync a suspended transaction between two nodes using t_suspend() from the TMX module and DMQ (or similar module)? More specifically, can I suspend it on one node and resume the transaction on another in a cluster?

4 6

sys.exit() in app_python3s
by Denys Pozniak 26 Sep '24

26 Sep '24

Hello! I'm planning to rewrite a native Kamailio script in KEMI Python, and I want to understand the differences between app_python3 vs app_python3s in the context of using sys.exit(). My native script uses the exit function in deep nested route[] blocks quite a bit and would like to keep this logic in KEMI without using return codes on top (return -255) towards ksr_request_route() Using sys.exit() in app_python3s gives an error in syslog (although the interpreter does not crash and everything works correctly), in app_python3 no such error ERROR: app_python3s [apy3s_kemi.c:146]: apy3s_exec_func(): error exception occurred How safe is it to use sys.exit() in this case? -- BR, Denys Pozniak

1 0

Re: About error running PCSCF
by LUIS FELIPE RAMOS TORRES 25 Sep '24

25 Sep '24

Dear community: If someone runs this tutorial successfully for dockerized open5GS, please I appreciate it. Web page of OPEN 5GS: [image: image.png] The APNs were created in the docker_host:8080/docs [image: image.png] When i ran the command: docker compose -f srsenb.yaml up - d && docker container attach srsenb [image: image.png] [image: image.png] I created 2 APNs on the phones, but when I select one of them (internet), I have access to the Internet. About VOLTE, I don't know how to try or maybe I must change to the other APN (ims). [image: image.png] Please let me know, I only have the problem with VOLTE service and how to try it. Best wishes, Luis El mar, 24 sept 2024 a las 19:23, LUIS FELIPE RAMOS TORRES (< luis.ramost(a)pucp.edu.pe>) escribió: > > Sorry if I replied to you again. > > If you can explain to me why we need to create 2 APNs, 1 for the ims and > other for the internet. I created the subscriber on Open5GS. > > [image: image.png] > > > I also create the apns. I attached one of them to this email. > [image: image.png] > > I do everything that you put on github, but when I run the command for the > OTA, my cell phone registers to the network, and when I create manually the > 2 APNS, I select the "internet" and have access, but I don't know how to > configure VOLTE. The model is a Huawei p30 lite, and also I don't know if > it supports VOLTE. > > [image: image.png] > [image: image.png] > > Cellular connected to the network: > [image: image.png] > > > Also in this tutorial ( > https://open5gs.org/open5gs/docs/tutorial/03-VoLTE-dockerized/ ), I see > that you use an app called CoIMS for VoLTE. Is that enabled for that > service or what to do exactly? > > Sorry again for my replies and comments. > > Luis > > > El mar, 24 sept 2024 a las 17:04, LUIS FELIPE RAMOS TORRES (< > luis.ramost(a)pucp.edu.pe>) escribió: > >> Hi Supreeth: >> >> My question is if the MSISDN is necessary to add to the parameters, >> because when I configure the SIM card, I don't generate any for that value. >> Or maybe to make a call. >> >> And if I have an error in making the call for VOLTE, maybe it would be a >> problem of a configuration, or maybe the phones. How to know that? >> >> Luis >> >> El mar, 24 sept 2024 a las 2:48, Supreeth Herle (<herlesupreeth(a)gmail.com>) >> escribió: >> >>> Hi Luis, >>> >>> In the next screenshot, I don't know if the next command (docker compose >>>> -f srsenb-yaml up ...) must run in another terminal window or in the same >>>> as I run the first command. >>> >>> >>> In another terminal window. >>> >>> I will not use the RF simulated because it would be physical and I have >>>> the LIMESDR hardware. Also the UE with 4G sim cards. >>> >>> >>> Run only the one which says OTA (over-the-air) >>> >>> And next of the following steps, when I run the others and add the >>>> subscriber, the apn, the ims, and the SIM parameters. What is the next >>>> step? Try the solution by making a call? >>>> >>> >>> yes >>> >>> BR, >>> Supreeth >>> >>> >>> >>> On Tue, 24 Sept 2024 at 02:35, LUIS FELIPE RAMOS TORRES < >>> luis.ramost(a)pucp.edu.pe> wrote: >>> >>>> Hi Supreeth: >>>> >>>> I do nothing but finally it runs successfully. But I have a question. I >>>> am running now the command of the screenshot (docker compose -f >>>> 4g-volte-deploy.yaml up) and I have the next output: >>>> >>>> [image: image.png] >>>> >>>> >>>> In the next screenshot, I don't know if the next command (docker >>>> compose -f srsenb-yaml up ...) must run in another terminal window or in >>>> the same as I run the first command. >>>> >>>> [image: image.png] >>>> I will not use the RF simulated because it would be physical and I have >>>> the LIMESDR hardware. Also the UE with 4G sim cards. >>>> >>>> And next of the following steps, when I run the others and add the >>>> subscriber, the apn, the ims, and the SIM parameters. What is the next >>>> step? Try the solution by making a call? >>>> >>>> Please if you can help me. >>>> >>>> Best wishes, >>>> Luis >>>> >>>> >>>> El dom, 22 sept 2024 a las 1:46, Supreeth Herle (< >>>> herlesupreeth(a)gmail.com>) escribió: >>>> >>>>> Hi Luis, >>>>> >>>>> Please raise an issue in >>>>> https://github.com/herlesupreeth/docker_open5gs >>>>> <https://github.com/herlesupreeth/docker_open5gs/tree/master> repo. I >>>>> will try to resolve it. >>>>> >>>>> I just ran the same command (after removing the image built locally) >>>>> and that step passed. So I would suggest to check your network connection >>>>> or DNS settings of the machine on which docker_open5gs is cloned. >>>>> >>>>> WARN[0000] /home/sherle/docker_open5gs/4g-volte-deploy.yaml: the >>>>>> attribute `version` is obsolete, it will be ignored, please remove it to >>>>>> avoid potential confusion >>>>>> [+] Building 19.0s (8/8) FINISHED >>>>>> >>>>>> >>>>>> >>>>>> docker:default >>>>>> => [osmohlr internal] load build definition from Dockerfile >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => => transferring dockerfile: 2.10kB >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => WARN: JSONArgsRecommended: JSON arguments recommended for CMD to >>>>>> prevent unintended behavior related to OS signals (line 41) >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => [osmohlr internal] load metadata for >>>>>> docker.io/library/debian:bullseye >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => [osmohlr internal] load .dockerignore >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => => transferring context: 2B >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => CACHED [osmohlr 1/3] FROM docker.io/library/debian:bullseye >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => [osmohlr 2/3] RUN apt-get update && apt-get -y install wget >>>>>> gnupg telnet >>>>>> >>>>>> >>>>>> 14.1s >>>>>> => [osmohlr 3/3] RUN wget >>>>>> https://downloads.osmocom.org/packages/osmocom:/latest/Debian_11/Release.key >>>>>> && mv Release.key /etc/apt/trusted.gpg.d/osmocom-latest.asc && echo >>>>>> "deb [signed-by=/etc/apt/trusted.gpg.d/osmocom-latest.asc] >>>>>> https://downloads.osmocom.org/packages/osmocom:/latest/Debian_11/ >>>>>> ./" > /etc/apt/sources.list.d/osmocom-latest.list && apt 4.5s >>>>>> => [osmohlr] exporting to image >>>>>> >>>>>> >>>>>> >>>>>> 0.2s >>>>>> => => exporting layers >>>>>> >>>>>> >>>>>> >>>>>> 0.2s >>>>>> => => writing image >>>>>> sha256:95227b65ff159ba4eab30f2caa460163b76895703c8cc73f41735d1ef4ef6dfc >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => => naming to docker.io/library/docker_osmohlr >>>>>> >>>>>> >>>>>> >>>>>> 0.0s >>>>>> => [osmohlr] resolving provenance for metadata file >>>>> >>>>> >>>>> BR, >>>>> Supreeth >>>>> >>>>> On Sat, 21 Sept 2024 at 20:42, LUIS FELIPE RAMOS TORRES < >>>>> luis.ramost(a)pucp.edu.pe> wrote: >>>>> >>>>>> Hello: >>>>>> >>>>>> I don't know if somebody can successfully run this tutorial for a >>>>>> solution of open5gs with kamailio using docker. I have the next error after >>>>>> the command: >>>>>> >>>>>> docker compose -f 4g-volte-deploy.yaml build >>>>>> and the output is: >>>>>> [image: image.png] >>>>>> >>>>>> [image: image.png] >>>>>> >>>>>> [image: image.png] >>>>>> >>>>>> I am running on Ubuntu 22.04, I also don't know if it is the same as >>>>>> Debian 11 (in the link of the download). Or how can I solve it? >>>>>> >>>>>> Luis >>>>>> >>>>>> El mié, 18 sept 2024 a las 19:30, LUIS FELIPE RAMOS TORRES (< >>>>>> luis.ramost(a)pucp.edu.pe>) escribió: >>>>>> >>>>>>> Hello >>>>>>> >>>>>>> Sorry for writing again, I followed your repository on github: >>>>>>> https://github.com/herlesupreeth/docker_open5gs/tree/master, but I >>>>>>> don't know how to build srslte: >>>>>>> >>>>>>> I have the next error when I build the srslte: docker build >>>>>>> --no-cache --force-rm -t docker_srslte . >>>>>>> >>>>>>> I don't know if it is updated, maybe also I run it on a VM, and >>>>>>> specifically in the "git checkout ......" I had the error. >>>>>>> >>>>>>> [image: image.png] >>>>>>> Luis >>>>>>> >>>>>>> El mar, 17 sept 2024 a las 1:45, Supreeth Herle (< >>>>>>> herlesupreeth(a)gmail.com>) escribió: >>>>>>> >>>>>>>> Hello Luis, >>>>>>>> >>>>>>>> When I use the github of RTPEngine, the master branch, when I run >>>>>>>>> the command dpkg-checkbuildeps >>>>>>>> >>>>>>>> >>>>>>>> I would not recommend installing the master branch of RTPEngine, >>>>>>>> rather use the tag mr9.4.1 and install on Ubuntu 20.04. You can find all >>>>>>>> dependencies to install here - >>>>>>>> https://github.com/herlesupreeth/docker_open5gs/blob/master/rtpengine/Docke… >>>>>>>> >>>>>>>> If you just want a working setup of IMS + RTPEngine I would highly >>>>>>>> recommend using https://github.com/herlesupreeth/docker_open5gs >>>>>>>> repo. You can get the setup running with few steps. >>>>>>>> >>>>>>>> BR, >>>>>>>> Supreeth >>>>>>>> >>>>>>>> On Tue, 17 Sept 2024 at 03:06, LUIS FELIPE RAMOS TORRES < >>>>>>>> luis.ramost(a)pucp.edu.pe> wrote: >>>>>>>> >>>>>>>>> Hello Supreeth: >>>>>>>>> >>>>>>>>> Sorry if I reply to the message, but I am trying to install the >>>>>>>>> tutorials, but I've got the same error. When I use the github of RTPEngine, >>>>>>>>> the master branch, when I run the command dpkg-checkbuildeps, I get errors. >>>>>>>>> >>>>>>>>> If you would have an updated tutorial to run successfully and also >>>>>>>>> recommend the Ubuntu version to avoid more errors, please it would help me. >>>>>>>>> >>>>>>>>> Luis >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> El lun, 16 sept 2024 a las 10:32, Supreeth Herle (< >>>>>>>>> herlesupreeth(a)gmail.com>) escribió: >>>>>>>>> >>>>>>>>>> Hello Luis, >>>>>>>>>> >>>>>>>>>> I am the author of the tutorial here - >>>>>>>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. That >>>>>>>>>> tutorial is also not up-to-date. However, you can use the latest master >>>>>>>>>> branch of kamailio, along with instructions here (installing dependencies >>>>>>>>>> etc) - >>>>>>>>>> https://github.com/herlesupreeth/docker_open5gs/blob/master/ims_base/Docker… >>>>>>>>>> and configuration files from master branch of this repo >>>>>>>>>> https://github.com/herlesupreeth/Kamailio_IMS_Config . >>>>>>>>>> >>>>>>>>>> As pointed out earlier, rtpproxy is not needed. It was a step I >>>>>>>>>> added to test out VoIP (not IMS) calling. All you need is RTPEngine to test >>>>>>>>>> out IMS. >>>>>>>>>> >>>>>>>>>> Hope it helps. >>>>>>>>>> >>>>>>>>>> BR, >>>>>>>>>> Supreeth >>>>>>>>>> >>>>>>>>>> On Mon, 16 Sept 2024 at 17:22, LUIS FELIPE RAMOS TORRES via >>>>>>>>>> sr-users <sr-users(a)lists.kamailio.org> wrote: >>>>>>>>>> >>>>>>>>>>> Hello Henning: >>>>>>>>>>> >>>>>>>>>>> Is this tutorial updated? >>>>>>>>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ >>>>>>>>>>> >>>>>>>>>>> I don´t know about it. And also I guess that it is installed on >>>>>>>>>>> Ubuntu 18 version. >>>>>>>>>>> >>>>>>>>>>> Cheers, >>>>>>>>>>> Luis >>>>>>>>>>> >>>>>>>>>>> El lun, 16 sept 2024 a las 1:32, Henning Westerholt (< >>>>>>>>>>> hw(a)gilawa.com>) escribió: >>>>>>>>>>> >>>>>>>>>>>> Hello, >>>>>>>>>>>> >>>>>>>>>>>> the tutorial is using a Kamailio version that its long end of >>>>>>>>>>>> life (5.3.x). >>>>>>>>>>>> >>>>>>>>>>>> Some of the instructions given there are also a bit odd, like >>>>>>>>>>>> first installing rtpproxy, and then replacing it with rtpengine. >>>>>>>>>>>> >>>>>>>>>>>> Maybe you can reach out to the author of the tutorial. You need >>>>>>>>>>>> to understand how the individual components work together in the end to be >>>>>>>>>>>> able to debug it. Maybe somebody else figured out on this list how to do it >>>>>>>>>>>> for an updated Kamailio. >>>>>>>>>>>> >>>>>>>>>>>> Cheers, >>>>>>>>>>>> >>>>>>>>>>>> Henning >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> -- >>>>>>>>>>>> Henning Westerholt - https://skalatan.de/blog/ >>>>>>>>>>>> Kamailio services - https://gilawa.com >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> > -----Original Message----- >>>>>>>>>>>> > From: luis.ramost--- via sr-users < >>>>>>>>>>>> sr-users(a)lists.kamailio.org> >>>>>>>>>>>> > Sent: Montag, 16. September 2024 02:14 >>>>>>>>>>>> > To: sr-users(a)lists.kamailio.org >>>>>>>>>>>> > Cc: luis.ramost(a)pucp.edu.pe >>>>>>>>>>>> > Subject: [SR-Users] About error running PCSCF >>>>>>>>>>>> > >>>>>>>>>>>> > Hi community: >>>>>>>>>>>> > >>>>>>>>>>>> > I am trying to run the Kamailio IMS, but I have the next >>>>>>>>>>>> error in the module >>>>>>>>>>>> > PCSCF >>>>>>>>>>>> > >>>>>>>>>>>> > I installed the Kamailio on a VM Machine, with UBuntu 18.04 >>>>>>>>>>>> > >>>>>>>>>>>> > Commands: >>>>>>>>>>>> > $ sudo -i >>>>>>>>>>>> > # mkdir -p /var/run/kamailio_pcscf >>>>>>>>>>>> > #kamailio -f /etc/kamailio_pcscf/kamailio_pcscf.cfg -P >>>>>>>>>>>> /kamailio_pcscf.pid -DD >>>>>>>>>>>> > -E -e >>>>>>>>>>>> > >>>>>>>>>>>> > Output: >>>>>>>>>>>> > loading modules under config path: >>>>>>>>>>>> > >>>>>>>>>>>> /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kama >>>>>>>>>>>> > >>>>>>>>>>>> ilio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux- >>>>>>>>>>>> > gnu/kamailio/modules/:/usr/local/lib64/kamailio/modules >>>>>>>>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: >>>>>>>>>>>> set_mod_param_regex(): >>>>>>>>>>>> > parameter <delete_delay> of type <2:int> not found in module >>>>>>>>>>>> > <ims_registrar_pcscf> >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 399, column >>>>>>>>>>>> 51: Can't set >>>>>>>>>>>> > module parameter >>>>>>>>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: >>>>>>>>>>>> set_mod_param_regex(): >>>>>>>>>>>> > parameter <ipsec_preferred_ealg> of type <1:string> not found >>>>>>>>>>>> in module >>>>>>>>>>>> > <ims_ipsec_pcscf> >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 417, column >>>>>>>>>>>> 59: Can't set >>>>>>>>>>>> > module parameter >>>>>>>>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: >>>>>>>>>>>> set_mod_param_regex(): >>>>>>>>>>>> > parameter <recv_mode> of type <2:int> not found in module >>>>>>>>>>>> <ims_qos> >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 435, column >>>>>>>>>>>> 35: Can't set >>>>>>>>>>>> > module parameter >>>>>>>>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: >>>>>>>>>>>> set_mod_param_regex(): >>>>>>>>>>>> > parameter <dialog_direction> of type <2:int> not found in >>>>>>>>>>>> module <ims_qos> >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 436, column >>>>>>>>>>>> 42: Can't set >>>>>>>>>>>> > module parameter >>>>>>>>>>>> > 0(470) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks >>>>>>>>>>>> array size 16 >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/register.cfg, line 260, column >>>>>>>>>>>> 42: unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_create (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/register.cfg, line 264, column >>>>>>>>>>>> 77: unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/mo.cfg, line 114, column 31: >>>>>>>>>>>> unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/mo.cfg, line 180, column 33: >>>>>>>>>>>> unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/mt.cfg, line 11, column 34: >>>>>>>>>>>> unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. >>>>>>>>>>>> parser: failed to find >>>>>>>>>>>> > command ipsec_forward (params 2) >>>>>>>>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): >>>>>>>>>>>> parse error in config >>>>>>>>>>>> > file /etc/kamailio_pcscf/route/mt.cfg, line 99, column 42: >>>>>>>>>>>> unknown >>>>>>>>>>>> > command, missing loadmodule? >>>>>>>>>>>> > >>>>>>>>>>>> > ERROR: bad config file (10 errors) >>>>>>>>>>>> > 0(470) WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): >>>>>>>>>>>> > WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: >>>>>>>>>>>> > cdp_avp_destroy(226) >>>>>>>>>>>> > 0(470) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer >>>>>>>>>>>> child stopping >>>>>>>>>>>> > ... >>>>>>>>>>>> > 0(470) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... >>>>>>>>>>>> CDiameterPeer child >>>>>>>>>>>> > stopped >>>>>>>>>>>> > 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: >>>>>>>>>>>> > mod_destroy(): Error destroying spi generator >>>>>>>>>>>> > 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:313]: >>>>>>>>>>>> > mod_destroy(): Error destroying port generator >>>>>>>>>>>> > >>>>>>>>>>>> > I guided form this tutorial: >>>>>>>>>>>> https://ryantheelder.github.io/blog/2023/VoLTE/ >>>>>>>>>>>> > >>>>>>>>>>>> > Please if somebody would help me, I appreciate it >>>>>>>>>>>> > >>>>>>>>>>>> > Finally my objective is to connect with srsRAN and open5GS to >>>>>>>>>>>> have VoLTE >>>>>>>>>>>> > service. >>>>>>>>>>>> > >>>>>>>>>>>> > Best wishes, >>>>>>>>>>>> > Luis >>>>>>>>>>>> > __________________________________________________________ >>>>>>>>>>>> > Kamailio - Users Mailing List - Non Commercial Discussions To >>>>>>>>>>>> unsubscribe >>>>>>>>>>>> > send an email to sr-users-leave(a)lists.kamailio.org >>>>>>>>>>>> > Important: keep the mailing list in the recipients, do not >>>>>>>>>>>> reply only to the >>>>>>>>>>>> > sender! >>>>>>>>>>>> > Edit mailing list options or unsubscribe: >>>>>>>>>>>> >>>>>>>>>>> __________________________________________________________ >>>>>>>>>>> Kamailio - Users Mailing List - Non Commercial Discussions >>>>>>>>>>> To unsubscribe send an email to >>>>>>>>>>> sr-users-leave(a)lists.kamailio.org >>>>>>>>>>> Important: keep the mailing list in the recipients, do not reply >>>>>>>>>>> only to the sender! >>>>>>>>>>> Edit mailing list options or unsubscribe: >>>>>>>>>>> >>>>>>>>>>

1 0

Re: [EXTERNAL] Re: SDP parsing fails when media stream is removed, there is no session-wide "c=" line and offending media stream omits all attributes
by Thomas Baumgartner 24 Sep '24

24 Sep '24

Hi Richard, thanks for your input. And i guess you are correct, when reading the sentence with the wording of the SDP RFC in mind attributes should only be "a=" lines. We will talk to the device vendor again with this new finding! Anyways, maybe it still makes sense to adjust the SDP parser for such scenarios, as it really doesn't make sense to require a "c=" line if the stream is being removed as you said. I will open an git issue aswell. Mit kommunikativen Grüßen I Best regards Thomas

2 1

SDP parsing fails when media stream is removed, there is no session-wide "c=" line and offending media stream omits all attributes
by Thomas Baumgartner 24 Sep '24

24 Sep '24

Hi, we have a problem with a device trying to send T.38 fax. During the T.38 ReInvite Kamailio fails to parse the SDP (via rtpengine_manage), and subsequently rtpengine is failing to do it's job. The problem seems to be that Kamailio is trying to find a media address, since there is no session-wide c= line present it goes to the media c= lines, but since the first media stream which was an audio stream is being removed (port 0) and can therefore omit all attributes (as per RFC 3264 8.2, or am i understanding this wrong?), the Kamailio SDP parser is failing, even if there is another c= line in the T.38 media stream. Here is the offending SDP from the ReInvite (IP changed): v=0 o=xmserver 1726638425 1726638427 IN IP4 169.254.1.1 s=xmserver t=0 0 m=audio 0 RTP/AVP 8 m=image 56002 udptl t38 c=IN IP4 169.254.1.1 a=sendrecv a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy And here is the corresponding log entry: <core> [core/parser/sdp/sdp.c:523]: parse_sdp_session(): can't find media IP in the message Can anyone confirm my findings? Is it possible to adjust the SDP parser for these kind of scenarios? Mit kommunikativen Grüßen I Best regards Thomas

3 3

t_next_contacts fails with 1 contact
by Igor Potjevlesch 24 Sep '24

24 Sep '24

Hello! I'm parsing 302 with, most of the time, multi contacts returned. I do it like this: get_redirects("*"); t_load_contacts(); and, finally, a test on !t_next_contacts(). This test fails when only one contact is present in 302. Is it the expected behaviour? From the documentation I understand that following t_load_contacts we must use t_next_contacts. If so, I believe that I shouldn't exit on !t_next_contacts() and do an additional test to see if there is, at least, one attempt to do toward the only contact returned? Thanks for your feedback. Regards, Igor. -- Cet e-mail a été vérifié par le logiciel antivirus d'Avast. www.avast.com

1 1

Honor Retry-After
by Knserbrave 23 Sep '24

23 Sep '24

Hi Community, Any advice on how to handle a Retry-After header locally without sleeping? I would like to suspend the transaction, schedule an event, wake up the transaction after the event timer triggers, and resume the transaction. I don't believe this is possible though.

3 4

Kamailio packages for 24.04.1 LTS (Noble Numbat)
by Leonid Fainshtein 23 Sep '24

23 Sep '24

Hello, Is it planned to release the Kamailio 5.8 deb packages for 24.04.1 LTS (Noble Numbat)? Best regards, Leonid

2 3

Pseudo Variable containing 'path' of main branch? / How to compose $du manually with path?
by Benoît Panizzon 23 Sep '24

23 Sep '24

Hi List I want to be able to parallel branch a call to multiple registered locations / multiple different AOR. So using the registrar lookup() function can not be used. I loop through all required AOR with reg_fetch_contacts, those could be registered via a proxy and therefore require to use Path: $ulc(aor=>path) in this case, contains the path to that destination. On all 'additional' branches added with append_branch() I can set the path using $branch(path). But i struggle with the main aka first branch on which I directly set: * $ru * $fs * $du From the documentation, if I would use the registrar lookup() function, then $du would correctly be set respecting the path for all contacts on an AOR. But who do I build $du manually with what I find in $ulc(aor=>path) when not using lookup()? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 6

Re: About error running PCSCF
by LUIS FELIPE RAMOS TORRES 21 Sep '24

21 Sep '24

Hello: I don't know if somebody can successfully run this tutorial for a solution of open5gs with kamailio using docker. I have the next error after the command: docker compose -f 4g-volte-deploy.yaml build and the output is: [image: image.png] [image: image.png] [image: image.png] I am running on Ubuntu 22.04, I also don't know if it is the same as Debian 11 (in the link of the download). Or how can I solve it? Luis El mié, 18 sept 2024 a las 19:30, LUIS FELIPE RAMOS TORRES (< luis.ramost(a)pucp.edu.pe>) escribió: > Hello > > Sorry for writing again, I followed your repository on github: > https://github.com/herlesupreeth/docker_open5gs/tree/master, but I don't > know how to build srslte: > > I have the next error when I build the srslte: docker build --no-cache > --force-rm -t docker_srslte . > > I don't know if it is updated, maybe also I run it on a VM, and > specifically in the "git checkout ......" I had the error. > > [image: image.png] > Luis > > El mar, 17 sept 2024 a las 1:45, Supreeth Herle (<herlesupreeth(a)gmail.com>) > escribió: > >> Hello Luis, >> >> When I use the github of RTPEngine, the master branch, when I run the >>> command dpkg-checkbuildeps >> >> >> I would not recommend installing the master branch of RTPEngine, rather >> use the tag mr9.4.1 and install on Ubuntu 20.04. You can find all >> dependencies to install here - >> https://github.com/herlesupreeth/docker_open5gs/blob/master/rtpengine/Docke… >> >> If you just want a working setup of IMS + RTPEngine I would highly >> recommend using https://github.com/herlesupreeth/docker_open5gs repo. >> You can get the setup running with few steps. >> >> BR, >> Supreeth >> >> On Tue, 17 Sept 2024 at 03:06, LUIS FELIPE RAMOS TORRES < >> luis.ramost(a)pucp.edu.pe> wrote: >> >>> Hello Supreeth: >>> >>> Sorry if I reply to the message, but I am trying to install the >>> tutorials, but I've got the same error. When I use the github of RTPEngine, >>> the master branch, when I run the command dpkg-checkbuildeps, I get errors. >>> >>> If you would have an updated tutorial to run successfully and also >>> recommend the Ubuntu version to avoid more errors, please it would help me. >>> >>> Luis >>> >>> >>> >>> El lun, 16 sept 2024 a las 10:32, Supreeth Herle (< >>> herlesupreeth(a)gmail.com>) escribió: >>> >>>> Hello Luis, >>>> >>>> I am the author of the tutorial here - >>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. That >>>> tutorial is also not up-to-date. However, you can use the latest master >>>> branch of kamailio, along with instructions here (installing dependencies >>>> etc) - >>>> https://github.com/herlesupreeth/docker_open5gs/blob/master/ims_base/Docker… >>>> and configuration files from master branch of this repo >>>> https://github.com/herlesupreeth/Kamailio_IMS_Config . >>>> >>>> As pointed out earlier, rtpproxy is not needed. It was a step I added >>>> to test out VoIP (not IMS) calling. All you need is RTPEngine to test out >>>> IMS. >>>> >>>> Hope it helps. >>>> >>>> BR, >>>> Supreeth >>>> >>>> On Mon, 16 Sept 2024 at 17:22, LUIS FELIPE RAMOS TORRES via sr-users < >>>> sr-users(a)lists.kamailio.org> wrote: >>>> >>>>> Hello Henning: >>>>> >>>>> Is this tutorial updated? >>>>> https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ >>>>> >>>>> I don´t know about it. And also I guess that it is installed on Ubuntu >>>>> 18 version. >>>>> >>>>> Cheers, >>>>> Luis >>>>> >>>>> El lun, 16 sept 2024 a las 1:32, Henning Westerholt (<hw(a)gilawa.com>) >>>>> escribió: >>>>> >>>>>> Hello, >>>>>> >>>>>> the tutorial is using a Kamailio version that its long end of life >>>>>> (5.3.x). >>>>>> >>>>>> Some of the instructions given there are also a bit odd, like first >>>>>> installing rtpproxy, and then replacing it with rtpengine. >>>>>> >>>>>> Maybe you can reach out to the author of the tutorial. You need to >>>>>> understand how the individual components work together in the end to be >>>>>> able to debug it. Maybe somebody else figured out on this list how to do it >>>>>> for an updated Kamailio. >>>>>> >>>>>> Cheers, >>>>>> >>>>>> Henning >>>>>> >>>>>> >>>>>> -- >>>>>> Henning Westerholt - https://skalatan.de/blog/ >>>>>> Kamailio services - https://gilawa.com >>>>>> >>>>>> >>>>>> >>>>>> > -----Original Message----- >>>>>> > From: luis.ramost--- via sr-users <sr-users(a)lists.kamailio.org> >>>>>> > Sent: Montag, 16. September 2024 02:14 >>>>>> > To: sr-users(a)lists.kamailio.org >>>>>> > Cc: luis.ramost(a)pucp.edu.pe >>>>>> > Subject: [SR-Users] About error running PCSCF >>>>>> > >>>>>> > Hi community: >>>>>> > >>>>>> > I am trying to run the Kamailio IMS, but I have the next error in >>>>>> the module >>>>>> > PCSCF >>>>>> > >>>>>> > I installed the Kamailio on a VM Machine, with UBuntu 18.04 >>>>>> > >>>>>> > Commands: >>>>>> > $ sudo -i >>>>>> > # mkdir -p /var/run/kamailio_pcscf >>>>>> > #kamailio -f /etc/kamailio_pcscf/kamailio_pcscf.cfg -P >>>>>> /kamailio_pcscf.pid -DD >>>>>> > -E -e >>>>>> > >>>>>> > Output: >>>>>> > loading modules under config path: >>>>>> > >>>>>> /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kama >>>>>> > ilio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux- >>>>>> > gnu/kamailio/modules/:/usr/local/lib64/kamailio/modules >>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): >>>>>> > parameter <delete_delay> of type <2:int> not found in module >>>>>> > <ims_registrar_pcscf> >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 399, column 51: >>>>>> Can't set >>>>>> > module parameter >>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): >>>>>> > parameter <ipsec_preferred_ealg> of type <1:string> not found in >>>>>> module >>>>>> > <ims_ipsec_pcscf> >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 417, column 59: >>>>>> Can't set >>>>>> > module parameter >>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): >>>>>> > parameter <recv_mode> of type <2:int> not found in module <ims_qos> >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 435, column 35: >>>>>> Can't set >>>>>> > module parameter >>>>>> > 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): >>>>>> > parameter <dialog_direction> of type <2:int> not found in module >>>>>> <ims_qos> >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 436, column 42: >>>>>> Can't set >>>>>> > module parameter >>>>>> > 0(470) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array >>>>>> size 16 >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/register.cfg, line 260, column 42: >>>>>> unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_create (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/register.cfg, line 264, column 77: >>>>>> unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/mo.cfg, line 114, column 31: unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/mo.cfg, line 180, column 33: unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/mt.cfg, line 11, column 34: unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: >>>>>> failed to find >>>>>> > command ipsec_forward (params 2) >>>>>> > 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse >>>>>> error in config >>>>>> > file /etc/kamailio_pcscf/route/mt.cfg, line 99, column 42: unknown >>>>>> > command, missing loadmodule? >>>>>> > >>>>>> > ERROR: bad config file (10 errors) >>>>>> > 0(470) WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): >>>>>> > WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: >>>>>> > cdp_avp_destroy(226) >>>>>> > 0(470) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child >>>>>> stopping >>>>>> > ... >>>>>> > 0(470) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer >>>>>> child >>>>>> > stopped >>>>>> > 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: >>>>>> > mod_destroy(): Error destroying spi generator >>>>>> > 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:313]: >>>>>> > mod_destroy(): Error destroying port generator >>>>>> > >>>>>> > I guided form this tutorial: >>>>>> https://ryantheelder.github.io/blog/2023/VoLTE/ >>>>>> > >>>>>> > Please if somebody would help me, I appreciate it >>>>>> > >>>>>> > Finally my objective is to connect with srsRAN and open5GS to have >>>>>> VoLTE >>>>>> > service. >>>>>> > >>>>>> > Best wishes, >>>>>> > Luis >>>>>> > __________________________________________________________ >>>>>> > Kamailio - Users Mailing List - Non Commercial Discussions To >>>>>> unsubscribe >>>>>> > send an email to sr-users-leave(a)lists.kamailio.org >>>>>> > Important: keep the mailing list in the recipients, do not reply >>>>>> only to the >>>>>> > sender! >>>>>> > Edit mailing list options or unsubscribe: >>>>>> >>>>> __________________________________________________________ >>>>> Kamailio - Users Mailing List - Non Commercial Discussions >>>>> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >>>>> Important: keep the mailing list in the recipients, do not reply only >>>>> to the sender! >>>>> Edit mailing list options or unsubscribe: >>>>> >>>>

1 0

Kamailio not listening to SIP despite setup OK
by Sergio Charrua 21 Sep '24

21 Sep '24

Hi all! got a client with 2 Kamailio servers using Pacemaker/Corosync with a VIP. Kamailio is setup with dispatcher, and has been working since 2018. This previous week, client migrated from a Cisco ASA firewall to a Fortinet F5 firewall. One of the Kamailio server is, apparently, working fine, receiving SIP invites and processing them correctly. When moving the VIP to the second Kamailio node, Kamailio is ignoring any SIP message that is not OPTIONS, despite configuration (kamailio.cfg) being exactly the same as in the node #1. Also, note that using SNGrep, we can see clearly that SIP messages are reaching the server, but Kamailio is just not processing any INVITE. In fact, I added an output to log file right at the very beginning of the main route and the line is added to log file ONLY if the SIP message is OPTIONS. When the SIP message is INVITE, nothing is processed on Kamailio's side. The listen parameter is set to 0.0.0.0 port 5060 and advertised IP set to the Public IP. Again, the main route is only : route{ xlog(. ......); } Kamailio version is 5.1.6 (old, I know....) but it has been working since Dec 2018! What could be the issue?! Why only OPTIONS are being , somehow, recognized by Kamailio and all other SIP messages just ignored? Also, to add some more weirdness to the mess, connecting an Asterisk server on the same network, and sending a call to Kamailio node #2, it works!! SIP messages of any kind are processed correctly by Kamailio on node #2. So, why doesn't it work with SIP messages received via Public IP address, even though the messages reach the server correctly (confirmed using SNGrep) ??? I spent 5h this afternoon trying to understand what is wrong, but I'm stuck .... and no clue so far, apart from suspecting of the new Gateway, which I have no access to configurations.... but if OPTIONS are received & processed OK, it wouldn't make sense that received INVITES would be ignored by Kamailio, right? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

6 7

Handling SIP Contact Header Changes and ACK Routing Issues
by XPTO 21 Sep '24

21 Sep '24

Hello community! I would like to ask for your help in solving a case. I thank you in advance for any collaboration. I have the following setup: - *Asterisk (internal)* → *Kamailio (Proxy)* → *SIP Provider (PSTN)* Both Asterisk and Kamailio are under my responsibility. *Problem:* When I initiate a call from Asterisk, I use Kamailio as the *outbound proxy*. The *INVITE* is sent to the SIP provider, and the dialogue proceeds normally until the provider responds with a *200 OK*, indicating the call was answered. However, the *200 OK* received from the provider contains a modification in the *Contact* header, introducing a new IP address into the dialogue. Presumably, this new IP is where the subsequent *ACK* should be sent. The issue arises when Kamailio receives this *200 OK*: it overwrites the *Contact* field, replacing the received IP with its own LAN IP, used for communication with Asterisk. As a result, Asterisk sends the *ACK* back to Kamailio, which, instead of forwarding it to the new IP from the *Contact* in the *200 OK*, sends it to the SIP provider at the same address used for the initial *INVITE*. Since the *ACK* does not reach the IP expected by the provider (the new IP introduced in the *Contact*), the provider continues to send repeated *200 OK* responses, until the call is disconnected due to a timeout, as a valid *ACK* was never received. I looked for conceptual answers in *RFC 3261* to support my conclusions, but I couldn’t clearly relate the information there to the behavior observed in this scenario. *Key Questions:* 1. *Should Kamailio forward the ACK to the new IP provided in the Contact header of the 200 OK?* 2. *Is the SIP provider correct in expecting the ACK to be sent to a different address from the original one used in the INVITE?* 3. *How can I configure Kamailio to ensure the dialogue proceeds correctly and the ACK is sent to the correct IP?*

2 1

Active/Active Registrar Design
by Knserbrave 20 Sep '24

20 Sep '24

Hi Community, Is it a common experience for architects to encounter issues with active/active registrar designs? Specifically, when clients are the UAS of a dialog? Some user agents accept INVITEs from any source once registered while others are not as accepting. There are ways around this using 302 redirect, but there seem to also be ways per RFC to signal trusted nodes to clients. RFC 3608 seems to be an excellent way to address this issue by communicating trustworthy nodes to the client upon registration. However, the Service-Route header seems to give options to the client rather than communicate trust. No where in the RFC does trust seem to be mentioned in its design, however one could imply it should be deduced by the presence of a Service-Route header. Nonetheless, even if it should communicate trust, it doesn't seem to do so in all cases so the more logical approach to active/active is signal via a 302 redirect where the registration lives, handling INVITE transactions on the server-side instead of client-side. I'm curious to hear if there is any advice on from the community overcoming trust issues with user agents in an active/active registrar design.

3 3

sni:uknown error in logs
by Mahmood Alkhalil 20 Sep '24

20 Sep '24

Hello Kamailio! I'm etting up a kamailio server where it will receive STIP TLS connections from Zoom. kamailio is closing TLS connections with error stating "SSL routines::no shared cipher (sni: unknown)" as below Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: tls [tls_server.c:270]: tls_complete_init(): Using initial TLS domain TLSs<default> (dom 0x7fbcd1e9dac8 ctx 0x7fbcd2229258 sn []) Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: tls [tls_domain.c:1018]: tls_server_name_cb(): SSL_get_servername returned NULL: return SSL_TLSEXT_ERR_NOACK Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: <core> [core/tcp_main.c:2845]: tcpconn_do_send(): sending... Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: <core> [core/tcp_main.c:2881]: tcpconn_do_send(): after real write: c= 0x7fbcd3cb85d0 n=7 fd=8 Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) DEBUG: <core> [core/tcp_main.c:2882]: tcpconn_do_send(): buf= Sep 18 13:28:02 dalia kamailio[18529]: [3B blob data] Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) ERROR: tls [tls_server.c:1312]: tls_h_read_f(): protocol level error Sep 18 13:28:02 dalia kamailio[18529]: 9(18529) ERROR: tls [tls_util.h:49]: tls_err_ret(): TLS accept:error:0A0000C1:SSL routines::no shared cipher (sni: unknown) did a tcpdump trace to check the ciphers Zoom are using in the TLS client hello, and there are 4 and are supported by openssl on TLSv1.2, BUT the reis no server_name extension in the client hello. is this related to kamailio refusing the connection because there is no server_name in the client hello or something else?, if yes can it be forced to accept TLS connection without server_name specified ? my tls.cfg file is below [server:default] method = TLSv1.2 verify_certificate = no require_certificate = no private_key = /etc/kamailio/key.pem certificate = /etc/kamailio/certificate.pem ca_list = /etc/ssl/certs/ca-certificates.crt ca_path = /etc/ssl/certs [client:default] method = TLSv1.2+ verify_certificate = no require_certificate = no

2 3

Accessing User-Agent from usrloc at runtime.
by Knserbrave 19 Sep '24

19 Sep '24

Hi Kamailio community. There are a few fields in the usrloc cache I am interested in accessing at runtime to influence routing decisions. I can use SQLops to achieve the same functionality, however I'm interested in minimizing DB operations. I would like to know if there is a generally accepted approach to retrieving this information from the memory-hosted usrloc cache rather than dipping into the database.

3 6

Re: Handling SIP Contact Header Changes and ACK Routing Issues
by Knserbrave 19 Sep '24

19 Sep '24

2 1

String Transformation to evaluate first part of header as key=value parameter?
by Benoit Panizzon 19 Sep '24

19 Sep '24

Hi List I have to parse a customer header: P-CH-Origin: intl=example.com;foo=bar;boo=whatever I'm interested in the value of the intl parameter. Unfortunately $(hdr(P-CH-Origin){param.value,intl}) does not seem to work, if this is the first parameter. Is there a built in function to do this? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 2

How to remove my email from Mailing list
by me.projects＠yahoo.com 19 Sep '24

19 Sep '24

Hi, I want to remove my email from mailing list. Thanks, ForTheGuru

2 1

Being added to the email marketing database of a 'kamailio developing' company after posting to list?
by Benoit Panizzon 18 Sep '24

18 Sep '24

Hi Today I received two marketing emails inviting me to the UCX Expo in London from I company I was not aware having any business relationship with. After searching for that company name in my email archive, I stumbled over a post from them in this mailing list. Just out of curiosity, have others on this list also received this invitation and in the past more marketing emails from the same source? I guess posting to this list does not imply, I have given permission to reader of this list to store my personal data in a marketing database of their employer and use my data for marketing purposes, right? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 2

Handling SIP Contact Header Changes and ACK Routing Issues
by Josimar Rocha 17 Sep '24

17 Sep '24

Hello community! I would like to ask for your help in solving a case. I thank you in advance for any collaboration. I have the following setup: Asterisk (internal) → Kamailio (Proxy) → SIP Provider (PSTN) Both Asterisk and Kamailio are under my responsibility. Problem: When I initiate a call from Asterisk, I use Kamailio as the outbound proxy. The INVITE is sent to the SIP provider, and the dialogue proceeds normally until the provider responds with a 200 OK, indicating the call was answered. However, the 200 OK received from the provider contains a modification in the Contact header, introducing a new IP address into the dialogue. Presumably, this new IP is where the subsequent ACK should be sent. The issue arises when Kamailio receives this 200 OK: it overwrites the Contact field, replacing the received IP with its own LAN IP, used for communication with Asterisk. As a result, Asterisk sends the ACK back to Kamailio, which, instead of forwarding it to the new IP from the Contact in the 200 OK, sends it to the SIP provider at the same address used for the initial INVITE. Since the ACK does not reach the IP expected by the provider (the new IP introduced in the Contact), the provider continues to send repeated 200 OK responses, until the call is disconnected due to a timeout, as a valid ACK was never received. I looked for conceptual answers in RFC 3261 to support my conclusions, but I couldn’t clearly relate the information there to the behavior observed in this scenario. Key Questions: Should Kamailio forward the ACK to the new IP provided in the Contact header of the 200 OK? Is the SIP provider correct in expecting the ACK to be sent to a different address from the original one used in the INVITE? How can I configure Kamailio to ensure the dialogue proceeds correctly and the ACK is sent to the correct IP?

1 0

Handling SIP Contact Header Changes and ACK Routing Issues
by Josimar Rocha 17 Sep '24

17 Sep '24

Hello community! I would like to ask for your help in solving a case. I thank you in advance for any collaboration. I have the following setup: Asterisk (internal) → Kamailio (Proxy) → SIP Provider (PSTN) Both Asterisk and Kamailio are under my responsibility. Problem: When I initiate a call from Asterisk, I use Kamailio as the outbound proxy. The INVITE is sent to the SIP provider, and the dialogue proceeds normally until the provider responds with a 200 OK, indicating the call was answered. However, the 200 OK received from the provider contains a modification in the Contact header, introducing a new IP address into the dialogue. Presumably, this new IP is where the subsequent ACK should be sent. The issue arises when Kamailio receives this 200 OK: it overwrites the Contact field, replacing the received IP with its own LAN IP, used for communication with Asterisk. As a result, Asterisk sends the ACK back to Kamailio, which, instead of forwarding it to the new IP from the Contact in the 200 OK, sends it to the SIP provider at the same address used for the initial INVITE. Since the ACK does not reach the IP expected by the provider (the new IP introduced in the Contact), the provider continues to send repeated 200 OK responses, until the call is disconnected due to a timeout, as a valid ACK was never received. I looked for conceptual answers in RFC 3261 to support my conclusions, but I couldn’t clearly relate the information there to the behavior observed in this scenario. Key Questions: *Should Kamailio forward the ACK to the new IP provided in the Contact header of the 200 OK? *Is the SIP provider correct in expecting the ACK to be sent to a different address from the original one used in the INVITE? *How can I configure Kamailio to ensure the dialogue proceeds correctly and the ACK is sent to the correct IP?

1 0

About error running PCSCF
by luis.ramost＠pucp.edu.pe 16 Sep '24

16 Sep '24

Hi community: I am trying to run the Kamailio IMS, but I have the next error in the module PCSCF I installed the Kamailio on a VM Machine, with UBuntu 18.04 Commands: $ sudo -i # mkdir -p /var/run/kamailio_pcscf #kamailio -f /etc/kamailio_pcscf/kamailio_pcscf.cfg -P /kamailio_pcscf.pid -DD -E -e Output: loading modules under config path: /usr/lib64/kamailio/modules_k/:/usr/lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux-gnu/kamailio/modules/:/usr/local/lib64/kamailio/modules 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <delete_delay> of type <2:int> not found in module <ims_registrar_pcscf> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 399, column 51: Can't set module parameter 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <ipsec_preferred_ealg> of type <1:string> not found in module <ims_ipsec_pcscf> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 417, column 59: Can't set module parameter 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <recv_mode> of type <2:int> not found in module <ims_qos> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 435, column 35: Can't set module parameter 0(470) ERROR: <core> [core/modparam.c:164]: set_mod_param_regex(): parameter <dialog_direction> of type <2:int> not found in module <ims_qos> 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio_pcscf.cfg, line 436, column 42: Can't set module parameter 0(470) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/register.cfg, line 260, column 42: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_create (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/register.cfg, line 264, column 77: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mo.cfg, line 114, column 31: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mo.cfg, line 180, column 33: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mt.cfg, line 11, column 34: unknown command, missing loadmodule? 0(470) ERROR: <core> [core/cfg.y:3399]: yyparse(): cfg. parser: failed to find command ipsec_forward (params 2) 0(470) CRITICAL: <core> [core/cfg.y:3539]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/route/mt.cfg, line 99, column 42: unknown command, missing loadmodule? ERROR: bad config file (10 errors) 0(470) WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) 0(470) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... 0(470) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:309]: mod_destroy(): Error destroying spi generator 0(470) ERROR: ims_ipsec_pcscf [ims_ipsec_pcscf_mod.c:313]: mod_destroy(): Error destroying port generator I guided form this tutorial: https://ryantheelder.github.io/blog/2023/VoLTE/ Please if somebody would help me, I appreciate it Finally my objective is to connect with srsRAN and open5GS to have VoLTE service. Best wishes, Luis

4 3

[Dialog module] Dialog Flag Removal and recommendation of dlg_manage()
by Xenofon Karamanos 16 Sep '24

16 Sep '24

Hello all, We would like to inform you about a recent PR<https://github.com/kamailio/kamailio/pull/3961> (https://github.com/kamailio/kamailio/pull/3961) that introduces changes related to the dialog flag handling. Summary of Changes: * Removal of dlg_flag: The dlg_flag has been removed from the dialog module. * Replace with call of dlg_manage(): We recommend using the function dlg_manage() as a replacement to handle the necessary functionality. Please refer to the updated documentation for more details. Affected Modules: 1. These modules only required updated documentation that references the use of dlg_flag instead dlg_manage(). * acc * qos * pua_dialoginfo * sst 1. Usage Changes (Dialog Flag Handling Removed): In these modules, the code related to reading and checking the dialog flag has been removed. Additionally, the setting of the message internal flag has been removed. If you need to track the dialog or call state, you MUST now call dlg_manage() explicitly in the kamailio config * call_control * cnxcc * mediaproxy * nat_traversal Please review the updated documentation for further guidance on the use of dlg_manage(). Feel free to reach out with any questions, concerns and report any issue due to this change. Best regards, Xenofon Karamanos

1 0

rtpengine module
by Daniel Donoghue 16 Sep '24

16 Sep '24

Hello, I am using rtpengine to encrypt and decrypt DTLS and SDES audio streams for my calls. I have a set of rtpengine instances running, configured using a database, such that it is easy to scale up on demand. This works well for initiating a call and balancing load across multiple instances but I want to be able to recover from rtpengine failure. I have looked at configuring rtpengine with a redis service but do not believe that to be helpful in this case as im using encrypted media and because the configuration of rtpengine with redis has a lot of overheads, limits scaling and requires non standard configuration of a redis cluster. I would like to be able to re-anchor media streams instead of relying on rtpengine to synchronise itself across a redis cluster. For this to work, i believe i would have to use the uac module to send a reINVITE and I speculate that i would need to use an xhttp event to trigger this remotely when a failure is detected. I am unclear, though, how to get to a specific call within kamailio from the xhttp route in order to do anything to it, if i even need to, and i am nuclear how to make the rtpengine module select a new node for sdp for the reinvite? How can i achieve my goal? Perhaps it would be possible to modify the rtpengine module itself to support generating a reinvite and selecting a new node on receipt of an rpc request? Hoping someone can help me find a way forward With every blessing, Daniel

3 6

About Rx AAR in PCSCF for mobile terminated call
by Liu Wing Hong 16 Sep '24

16 Sep '24

Hi, I am using kamailio 5.8.3 as an IMS-PCSCF. On handling SIP INVITE (and response), it sends Rx AAR to PCRF for bearer establishment. For mobile-terminated call, it was found that the call was dropped by handset within several seconds after ringing for some handsets. The corresponding pcap capture is attached for reference. From the pcap capture, it was found that in the Rx AAR request sent (packet 11), in the Media-Sub-Component Flow-Description, the "permit out from ip/port" and "permit in to ip/port" were set to 202.0.169.182 port 24428 instead of 10.51.0.15 port 31124. 202.0.169.182 port 24428 is the one in the incoming INVITE sdp from S-CSCF (packet 2); 10.51.0.15 port 31124 is the one in the processed INVITE sdp sent to UE (packet 5); Also, the sdp data of "offer" inside AAR was set to the sdp in the incoming INVITE from S-CSCF instead of the sdp in the processed INVITE sent to UE. Is that correct? Please advice. Thank you. Regards, Hong

1 0

Setting $du to websocket-secure
by James Browne 13 Sep '24

13 Sep '24

How can I set the destination URI for an INVITE to be a websocket-secure destination? Is it possible? Summary I've a proxy with tcp_connection_match=1, but websocket URIs always have transport=ws (never transport=wss) in them, so relaying a call to a WSS connection always fails. I tested running kamailio 6.0.0-dev2 compiled from a commit made this week. This proxy server uses nathelper rather than outbound module. Detail We know that "transport=ws" is used for both WS and WSS. I've a proxy server that receives an INVITE for a WSS destination, and this proxy supports both WS and WSS. This proxy server must have core parameter tcp_connection_match=1 set, and this leads the t_relay() to fail. When an INVITE comes, these are the steps. - The URI is something like sip:user@anonymous.invalid;alias=198.51.100.10~52833~6;transport=ws. - First handle_ruri_alias() removes the alias (which has ~6 in it, for wss) and sets the $du to something like sip:198.51.100.10:52833;transport=ws. - Then loose_route_preloaded() processes the Route header fields and forces the outbound socket to the TLS websocket one. - Then t_relay() fails to relay the INVITE and responds with 477 or 500. If, however, there's a non-TLS websocket connection open to the proxy, the INVITE would be erroneously relayed over that (using the wrong kamailio-side TCP port). I can go deeper with testing if required. I wonder whether this is a bug. James

3 12

Strange result: $stat(total_size) returns 0
by whliu＠citictel.com 12 Sep '24

12 Sep '24

Hi, I am using kamailio 5.8.3. I just found that strange result is returned from variable $stat(total_size). In the example script under "misc/examples/ims/pcscf/kamailio.cfg", there is a line, $var(used) = 1 - ($stat(free_size) / $stat(total_size)); The trace log said that $stat(total_size) was returning 0, and got a log like this ERROR: <core> [core/rvalue.c:1434]: long_longop2(): rv div by 0 Any idea on this? Please advise. Thank you. Regards, Hong

1 0

Strange result: $stat(total_size) returns 0
by whliu＠citictel.com 12 Sep '24

12 Sep '24

Hi, I am using kamailio 5.8.3. I just found that strange result is returned from variable $stat(total_size). In the example script under "misc/examples/ims/pcscf/kamailio.cfg", there is a line, $var(used) = 1 - ($stat(free_size) / $stat(total_size)); The trace log said that $stat(total_size) was returning 0, and got a log like this ERROR: <core> [core/rvalue.c:1434]: long_longop2(): rv div by 0 Any idea on this? Please advise. Thank you. Regards, Hong

1 0

SIP parse error for self-generated Websocket reply
by Sebastian Damm 12 Sep '24

12 Sep '24

Hello, We try to introduce a small websocket sidecar to our Kamailio servers, which runs on the same host. It connects to the regular TCP socket of our Kamailios, and in the XHTTP request route, we do just this: if path == "/sidecar": KSR.websocket.handle_handshake() return 1 On every new connect, Kamailio logs a parse error for the reply it generates. Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/parser/parse_fline.c:272]: parse_first_line(): parse_first_line: bad message (offset: 22) Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/parser/msg_parser.c:790]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols Sia: SIP/2.0/TCP 100.74.11.1:46422 Sec-WebSocket-Protocol: sip Upgrade: websocket Connection: upgrade Sec-WebSocket-Accept: 60dPbzcRW6up7z92kDlxkqL2WlY= Content-Length: 0 > Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/msg_translator.c:3402]: build_sip_msg_from_buf(): parsing failed I saw this error in other questions from a while ago, but without a solution. What could be the reason for this? Thanks in advance, Sebastian

2 2

Suspected memory leak in module ims_registrar_pcscf
by whliu＠citictel.com 09 Sep '24

09 Sep '24

Hi, I just tried to send numbers of SIP register to kamailio (release 5.8.2). It is reported that there are some memory leakage in shared memory after the expiry of all the registration sessions. After drill down the code. function save_pending(...) is called. (in ims_registrar_pcscf/save.c). Inside that function, it seems that "sec_verify_params" is not deallocated upon the function completes. Findings: - sec_verify_params uses the result from function cscf_get_security_verify(...) - Function cscf_get_security_verify(...) returns the result of function parse_sec_agree(...) - Function parse_sec_agree(...) returns "params" where "shm_malloc(...)" is called It seems that the memory pointed by 'sec_verify_params' is not further referenced by others. Should shm_free(...) be called when leaving function save_pending(...) in order to free the unused memory of 'sec_verify_params'? Please advice. Thank you. Regards, Hong

2 2

RFC: Support for icc or suncc compilers
by Daniel-Constantin Mierla 09 Sep '24

09 Sep '24

Hello, I was wondering if icc or suncc compilers are still used in the community, and by that if it still worth trying to support them via Makefiles and compile time flags (defines). Personally I haven't had to deal with them for more than 10-15 years, I have no access with a system offering them. Furthermore, I am not even sure if the code compiles with them. Maybe someone (or many) in the community use one or the other and confirm that they still worth to be kept. If not, maybe we can consider them for removing, it can be a good task to be done during Kamailio development meeting planned in Dusseldorf, Germany during November. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, October 8-10, 2024 -- asipto.com

1 0

Problem on uac_req via port
by whliu＠citictel.com 09 Sep '24

09 Sep '24

Hi, I just tried to use kamailio 5.8.3 as an IMS-PCSCF. After handling SIP registration, it seems that value of port in the "via" header in the request sent from PCSCF to UE is a bit strange. When using uac_req to send OPTIONS to UE, it is found that in the OPTIONS sent out from kamailio, the value of port inside the "via" header is set to port_pc (pcscf ipsec client port). Is this correct? Or should the via header port be set to port_ps (pcscf ipsec server port) instead? Also, in uac_req, is the value of via port header configurable? Please advise. Thank you. Regards, Hong

1 0

Error when passing task to an async task group
by elhar.mohamed＠gmail.com 06 Sep '24

06 Sep '24

Hello, I'm using async task group to handle some pourcentage of the traffic, while testing with traffic of 1200cps I got the following error : ERROR: {1910812149 INVITE 6785db25bc3f2478@ip} <core> [core/async_task.c:439]: async_task_group_push(): failed to pass the task [0x7f4000c3a5c0] to group [async_group_name]. conf: async_workers_group="name=async_group_name;workers=45;nonblock=1;usleep=100" If someone can help please?

1 0

Disable topos for a specific peering IP
by Denys Pozniak 06 Sep '24

06 Sep '24

Hello! I need to disable topos for one specific SIP trunk (in-out), it looks like it’s enough to use event_route with IP address filtering. But for some reason, the incoming INVITE from the peer still gets processed by topos and I also don’t see a mention of [msg-incoming] in the logs, only this: WARNING: <script>: [msg-outgoing] OPTIONS/you/1.1.1.1 Code snippet: loadmodule "topos.so" modparam("topos", "db_url", DBURL) modparam("topos", "contact_mode", 1) modparam("topos", "header_mode", 1) modparam("topos", "methods_noinitial", "OPTIONS,SUBSCRIBE,PUBLISH") modparam("topos", "dialog_expire", 7210) modparam("topos", "rr_update", 1) modparam("topos", "event_mode", 5) /* 1 - execute event_route[topos:msg-outgoing] 2 - execute event_route[topos:msg-sending] 4 - execute event_route[topos:msg-incoming] 8 - execute event_route[topos:msg-receiving] */ request_route { .... event_route[topos:msg-outgoing] { if ( $sndto(ip) == "1.1.1.1" ) { xlog("L_WARN","[msg-outgoing] $rm/$rU/$sndto(ip) \n"); drop; } } } event_route[topos:msg-incoming] { if ( $si == "1.1.1.1" ) { xlog("L_WARN","[msg-incoming] $rm/$rU/$si \n"); drop; } } # kamailio -v version: kamailio 5.7.1 (x86_64/linux) 1cf389-dirty -- BR, Denys Pozniak

6 9

t_check_trans in Lua
by nizam＠ctapps.com 06 Sep '24

06 Sep '24

Dear experts, Would appreciate if anyone can explain why in the given sample Lua script the 'Handle Retransmission' block checks the return code of t_check_trans (if KSR.tm.t_check_trans == 0 then), but in the Kamailio native config standard file if just calls the t_check_trans() function without checking the return code. What is the intention here and how does it work? Thank you in advance. Handle retransmission script from native config file: if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } Handle retransmission from sample Lua script: if not KSR.is_ACK() then if KSR.tmx.t_precheck_trans() > 0 then KSR.tm.t_check_trans(); return 1; end if KSR.tm.t_check_trans() == 0 then return 1; end end

2 2

secsipid and secsipid_proc module RPM build
by Володимир Іванець 04 Sep '24

04 Sep '24

Hello! I was wondering if someone was trying to build Kamailio RPM packages containing modules secsipid and secsipid_proc and faced the same problem. I compiled secsipidx library and copied secsipid.h, libsecsipid.h as well as libsecsipid.a files into the source directory of secsipid_proc module. And when I'm trying to build RPM packages, I get the following error: CC (gcc) [M secsipid_proc.so] secsipid_proc_mod.o secsipid_proc_mod.c:28:10: fatal error: secsipid.h: No such file or directory #include <secsipid.h> ^~~~~~~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: secsipid_proc_mod.o] Error 1 make[1]: *** [Makefile:508: modules] Error 1 make[1]: Leaving directory '/builddir/build/BUILD/kamailio-5.8.2/src' make: *** [Makefile:34: every-module] Error 2 Thanks a lot!

2 4

Kamailio v5.8.3 Released
by Daniel-Constantin Mierla 04 Sep '24

04 Sep '24

Hello, Kamailio SIP Server v5.8.3 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.3. For more details about version 5.8.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/09/kamailio-v5-8-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, October 8-10, 2024 -- asipto.com

1 0

Permissions module - automatic reload
by Sergio Charrua 04 Sep '24

04 Sep '24

Hi all, Just need some clarifications about the Permissions module: I currently use it with db_mode = 1 and was expecting that the parameter "trusted_cleanup_interval" would refresh the cache tables every X seconds, but apparently this is not the case. I was looking for a way for Kamailio to automatically reload the address table into memory every 5 minutes, for example, and I (mis)understood that "trusted_cleanup_interval" would do the trick, but it doesn't work as expected. Question: is there any command, besides kamcmd, to automatically reload the permissions table? I would like to avoid having to add a line in the crontab.... Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

1 0

Using a different OpenSSL
by David Cunningham 04 Sep '24

04 Sep '24

Hello, We have compiled openssl 3.0.9 from source because it's FIPS validated, and want to use it with Kamailio. The server also has the Ubuntu openssl 3.0.2 package installed. Does anyone know how we can tell Kamailio to use the openssl library in /opt/openssl/lib64, and how we can verify that it really is using it? Thanking you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

5 13

Releasing v5.8.3
by Daniel-Constantin Mierla 04 Sep '24

04 Sep '24

Hello, I am considering to release Kamailio v5.8.3 (out of branch 5.8) sometime next week, most likely on Wednesday, Sep 4, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

kamailio stri shaken configuration
by Vishal Pai 02 Sep '24

02 Sep '24

Hello All I need to integrate the kamailio with stri shaken with pem. What is the best way to achieve this? Thanks

3 2

app_mono module to be archived
by Victor Seva 31 Aug '24

31 Aug '24

Hi kamilians, we have the intention to archive the app_mono module in a few[0]. Most likely at the end of next week. If you use app_mono and you don't want to see it fade away now it's the time to rise your voice. Cheers, Victor [0] https://github.com/kamailio/kamailio/pull/3964

1 0

How to restrict Kamailio adding Local VIA header
by panda.paresh＠gmail.com 30 Aug '24

30 Aug '24

Hi All, I am trying to replay a request to the destination via Kamailio, and I am using t_relay_to_tcp for that , but in that case Kamailio is adding its own address as the top VIA header. Is there a way I can instruct Kamailio not to add local via header ?

3 4

http_async and tm
by alexis 30 Aug '24

30 Aug '24

Hello all, context first, we have an REST API that performs queries to external devices in the network (diameter to DRA's, REST to different servers) and based on n conditions returns the content for a Contact header to be used in a SIP 302. Now we're consuming this API with http_client (synchronously) and as there's no way to speed up the API (pipeline executions, delays on external api's etc etc) we're hitting a limit where all children become busy waiting for the API to answer. So i decided to move to http_async_client and started working on it on the lab with this first and base concept to test. request_route { #for testing purposes only if(is_method("ACK")){ exit; } $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = 500; $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; jansson_set("string", "event", "sip-routing", "$var(cre_query)"); xlog("L_INFO","API ASYNC ROUTING REQUEST: $var(cre_query)\n"); $http_req(body) = $var(cre_query); t_newtran(); http_async_query("http://192.168.86.128:8000/", "CRE_RESPONSE"); } http://192.168.86.128:8000/ receives the POST, randomly creates a delay between 0.5 and 1 second and responds (simulating the real api with an excess delay to probe the concept) Then route[CRE_RESPONSE] { if ($http_ok && $http_rs == 200) { xlog("L_INFO","CRE RESPONSE: $http_rb\n"); # for testing purpose, Contact content will be replaced from the received api response append_to_reply("Contact: <sip:1234@google.com>\r\n"); send_reply(302,"Moved Temporarily"); exit; } send_reply(500, "Internal error"); exit; } INVITE is received and processed, API is called, after API response, 302 is replied and then an ACK (ignored by now). Situation is that the 302 retransmitted 37 1519.846253067 192.168.86.34 → 192.168.86.128 SIP/SDP 585 Request: INVITE sip:service@192.168.86.128:5060 | 38 1519.848100380 192.168.86.128 → 192.168.86.34 SIP 318 Status: 100 Trying | 39 1520.094997642 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 40 1520.102323728 192.168.86.34 → 192.168.86.128 SIP 453 Request: ACK sip:service@192.168.86.128:5060 | 41 1520.591300933 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 42 1521.591061065 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 43 1523.591227956 192.168.86.128 → 192.168.86.34 SIP 407 Status: 302 Moved Temporarily | 18(24) DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7f6d79745dc0: SIP/2.0 3..., shmem=0x7f6d75187fd8: SIP/2.0 3 18(24) DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7f6d79745dc0: SIP/2.0 3..., shmem=0x7f6d75187fd8: SIP/2.0 3 18(24) DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7f6d79745dc0: SIP/2.0 3..., shmem=0x7f6d75187fd8: SIP/2.0 3 18(24) DEBUG: tm [timer.c:634]: wait_handler(): finished transaction: 0x7f6d75184cc8 (p:0x7f6d74f600c8/n:0x7f6d74f600c8) 18(24) DEBUG: tm [h_table.c:132]: free_cell_helper(): freeing transaction 0x7f6d75184cc8 from timer.c:643 Any help to avoid the retransmission and make the transaction just finish right after the 302 will be appreciated. regards

6 27

apply keepalive to only udp
by James Lipski 30 Aug '24

30 Aug '24

Hello, I wanted to know if this can be accomplished, or if I'm doing something wrong. What I'm hoping to accomplish is to have kamailio periodically send OPTIONS packets to endpoints that are connected over UDP only. As of right now, I currently have the following configured under usrloc: modparam("usrloc", "timer_interval", 5) modparam("usrloc", "timer_procs", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 0) modparam("usrloc", "ka_mode", 1) modparam("usrloc", "ka_method", "OPTIONS") modparam("usrloc", "ka_from", "sip:ping@sip-domain.com") modparam("usrloc", "ka_domain", "sip-domain.com") modparam("usrloc", "ka_timeout", 125) modparam("usrloc", "ka_interval", 60) With these parameters, all endpoints will be sent OPTIONS packets periodically as expected. When setting "ka_mode" to "4", as expected, endpoints registered with UDP only receive OPTIONS packets; which is what I wanted. The issue that I'm having with these parameters is that endpoints registered with anything other that UDP are triggering a contact-expired event after a little over 2 minutes. I have the registrar set to 3600 as an expiry; I'm assuming the contact is expiring due to the "ka_timeout" value. -------------------------- Is there a way to only have keep-alive packets/ and timeouts apply to endpoints registered on UDP? Thank you.

3 5

Kamailio integration with REST services
by Sergio Charrua 29 Aug '24

29 Aug '24

Hi all! I need to integrate a REST Service with Kamailio. Integration is easy, not a problem, but my concern is that the REST Services has a JWT token authentication required for each request, common to many (it not all) REST Services. So, it is required to POST for authentication (username and password) and once I get authentication done, and receive a JWT token, I will need to use this token in all other requests to the REST Service. The token is valid for 24h. My question is: - how to save the token for use during 24h, on all calls/sessions, without the need to authenticate on every call? - how to detect the token has expired and re-authenticate? I looked for some modules, but could not find any that suited me. I am thinking of executing a Python script for this, but concerned about latency and PDD.... Any thoughts? Any suggestions? Any known best-practice? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

6 12

kamailio randomly crashes
by Maharaja Azhagiah 28 Aug '24

28 Aug '24

Hi all, I have kamailio running in GKE (Google Kubernetes) and recently seeing kamailio crashes frequently and I couldn't find the reason for the crash kamailio version version: kamailio 5.4.9 (x86_64/linux) 7d022d flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 7d022d compiled on 14:26:30 Jul 10 2024 with gcc 8.3.0 kamailio core generation: Core was generated by `kamailio -e -E -DD -M 128 -m 2048'. Program terminated with signal SIGSEGV, Segmentation fault. #0 load_route_data_db (rd=0x7ca482a48840) at cr_db.c:416 416 if (rd->carriers[i]->domains[j]->sum_prob == 0.0) { bt: (gdb) bt #0 load_route_data_db (rd=0x7ca482a48840) at cr_db.c:416 #1 0x00007ca474c8cc50 in reload_route_data () at cr_data.c:178 #2 0x00007ca474cabfdc in cr_rpc_reload_routes (rpc=0x7ca4f5a32aa0 <func_param>, c=0x7ca4f5a329c0 <_jsonrpc_ctx_global>) at cr_rpc.c:59 #3 0x00007ca4f5a158a1 in jsonrpc_exec_ex (cmd=0x7ffd774f34b0, rpath=0x0) at jsonrpcs_mod.c:1425 #4 0x00007ca4f5a15ccd in jsonrpc_exec (msg=0x7ca4f641a9f8, cmd=0x7ca4f61bde20 "x\335\033\366\244|", s2=0x0) at jsonrpcs_mod.c:1445 #5 0x00005745eb3fcb19 in do_action (h=0x7ffd774f3dc0, a=0x7ca4f61f1a50, msg=0x7ca4f641a9f8) at core/action.c:1085 #6 0x00005745eb40b35c in run_actions (h=0x7ffd774f3dc0, a=0x7ca4f61f0468, msg=0x7ca4f641a9f8) at core/action.c:1584 #7 0x00007ca474b92f8d in timer_handler (ticks=96802941, tl=0x7ca47b2f7950, data=0x7ca47b2f7868) at timer.c:216 #8 0x00005745eb49e3c4 in slow_timer_main () at core/timer.c:1105 #9 0x00005745eb3a85bd in main_loop () at main.c:1765 #10 0x00005745eb3b34c3 in main (argc=8, argv=0x7ffd774f4688) at main.c:2942 bt full: (gdb) bt full #0 load_route_data_db (rd=0x7ca482a48840) at cr_db.c:416 res = 0x7ca4f636a4a8 row = 0x7ca4f642a7f8 i = 0 j = 110 ret = 56 tmp_carrier_data = 0x7ca479ff5068 query_str = {s = 0x7ca474cdd800 <query> "SELECT DISTINCT domain FROM carrierroute WHERE carrier=8", len = 56} tmp_scan_prefix = {s = 0x5745ed889f68 "13235791374", len = 11} tmp_rewrite_host = {s = 0x5745ed889f7a "sip.telnyx.com;transport=TLS", len = 28} tmp_rewrite_prefix = {s = 0x5745ed889f99 "", len = 0} tmp_rewrite_suffix = {s = 0x5745ed889f9a "", len = 0} tmp_host_name = {s = 0x40 <error: Cannot access memory at address 0x40>, len = 1965559808} tmp_reply_code = {s = 0x7ca474ccf2b0 <__func__.5843> "reload_route_data", len = 1959587872} tmp_comment = {s = 0x5745ed889f9b "IMPORTED - WTJHSX", len = 18} p_tmp_comment = 0x7ffd774f30b0 __func__ = "load_route_data_db" n = 0 #1 0x00007ca474c8cc50 in reload_route_data () at cr_data.c:178 old_data = 0x5745ed850150 new_data = 0x7ca482a48840 i = 22341 __func__ = "reload_route_data" #2 0x00007ca474cabfdc in cr_rpc_reload_routes (rpc=0x7ca4f5a32aa0 <func_param>, c=0x7ca4f5a329c0 <_jsonrpc_ctx_global>) at cr_rpc.c:59 __func__ = "cr_rpc_reload_routes" #3 0x00007ca4f5a158a1 in jsonrpc_exec_ex (cmd=0x7ffd774f34b0, rpath=0x0) at jsonrpcs_mod.c:1425 rpce = 0x7ca4f6425fb8 ctx = 0x7ca4f5a329c0 <_jsonrpc_ctx_global> ret = -1 nj = 0x0 val = {s = 0x5745ed845010 "cr.reload_routes", len = 16} scmd = {s = 0x7ca4f61bdd78 "{\"jsonrpc\": \"2.0\", \"method\": \"cr.reload_routes\", \"id\": 2}", len = 57} __func__ = "jsonrpc_exec_ex" #4 0x00007ca4f5a15ccd in jsonrpc_exec (msg=0x7ca4f641a9f8, cmd=0x7ca4f61bde20 "x\335\033\366\244|", s2=0x0) at jsonrpcs_mod.c:1445 scmd = {s = 0x7ca4f61bdd78 "{\"jsonrpc\": \"2.0\", \"method\": \"cr.reload_routes\", \"id\": 2}", len = 57} __func__ = "jsonrpc_exec" #5 0x00005745eb3fcb19 in do_action (h=0x7ffd774f3dc0, a=0x7ca4f61f1a50, msg=0x7ca4f641a9f8) at core/action.c:1085 ret = -5 v = 0 dst = {send_sock = 0x0, to = {s = {sa_family = 41744, sa_data = "6\366\244|\000\000\035u\271t\244|\000"}, sin = {sin_family = 41744, sin_port = 63030, sin_addr = {s_addr = 31908}, sin_zero = "\035u\271t\244|\000"}, sin6 = {sin6_family = 41744, sin6_port = 63030, sin6_flowinfo = 31908, sin6_addr = {__in6_u = {__u6_addr8 = "\035u\271t\244|\000\000\005u\271t\244|\000", __u6_addr16 = {29981, 29881, 31908, 0, 29957, 29881, 31908, 0}, __u6_addr32 = {1958311197, 31908, 1958311173, 31908}}}, sin6_scope_id = 2001680800}, sas = {ss_family = 41744, __ss_padding = "6\366\244|\000\000\035u\271t\244|\000\000\005u\271t\244|\000\000\240\071Ow\375\177\000\000/.:\365\244|\000\000\320:Ow\375\177\000\000\327\337j\353EW\000\000\320\071Ow\375\177\000\000\020@ \016\366\244|\000\000\030\245\066\366\244|\000\000\020@\016\366\244|\000\000\020:Ow\375\177\000\000\356u\236\365\244|\000\000\030\245\066\366\244|\000\000\000\000\000\000\000\000\000", __ss_align = 134217728}}, id = 130848392, send_flags = {f = 0, blst_imask = 0}, proto = 56 '8', proto_pad0 = 5 '\005', proto_pad1 = 36} tmp = 0x7ca4f641a9f8 "\344\032" new_uri = 0xed3 <error: Cannot access memory at address 0xed3> end = 0x7ca474b9751b "\r\n" crt = 0x2 <error: Cannot access memory at address 0x2> cmd = 0x7ca4f617b7d8 len = 0 user = 15 uri = {user = {s = 0x9774f3800 <error: Cannot access memory at address 0x9774f3800>, len = 621}, passwd = {s = 0x400000000 <error: Cannot access memory at address 0x400000000>, len = 1721781302}, host = {s = 0x0, len = -2147483648}, port = {s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, params = {s = 0x7ffd774f38e0 "09Ow\375\177", len = -167557169}, sip_params = { s = 0x7ca4f53704f0 <dialog_table_name> "l^5\365\244|", len = -164192016}, headers = {s = 0x1700000000 <error: Cannot access memory at address 0x1700000000>, len = 2001681040}, port_no = 14800, proto = 30543, type = 32765, flags = (unknown: 4130775280), transport = {s = 0x0, len = 2009201968}, ttl = {s = 0x6774f3a50 <error: Cannot access memory at address 0x6774f3a50>, len = -181058744}, user_param = { s = 0x17200000000 <error: Cannot access memory at address 0x17200000000>, len = 2099608304}, maddr = {s = 0x7ffd774f3960 "\035u\271t\244|", len = -344235896}, method = {s = 0x0, len = -2147483648}, lr = { --Type <RET> for more, q to quit, c to continue without paging-- s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, r2 = {s = 0x7ffd774f38d0 "`9Ow\375\177", len = -166931844}, gr = {s = 0x7ca4f53704f0 <dialog_table_name> "l^5\365\244|", len = -164192016}, transport_val = {s = 0x7ffd774f3960 "\035u\271t\244|", len = -181636538}, ttl_val = {s = 0x7ffd774f3930 "`9Ow\375\177", len = -167555326}, user_param_val = {s = 0x0, len = 2}, maddr_val = { s = 0x7ca4f60c1f77 <db_mysql_submit_query> "UH\211\345AWAVAUATSH\201\354\210", len = -166889584}, method_val = {s = 0x177d6038c8 <error: Cannot access memory at address 0x177d6038c8>, len = 2001681040}, lr_val = {s = 0xe00000000 <error: Cannot access memory at address 0xe00000000>, len = 1958311183}, r2_val = {s = 0x7ffd774f3960 "\035u\271t\244|", len = -164191464}, gr_val = { s = 0x7ffd774f39f0 "\373\377\377\377", len = -344973614}} next_hop = {user = {s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, passwd = {s = 0x7ffd774f3720 "\020\070Ow\375\177", len = -166975523}, host = { s = 0x7ffd774f3740 "\020\070Ow\375\177", len = -166887566}, port = {s = 0x7ca4f60431d0 <sql_str> "@@w\355EW", len = -164192016}, params = {s = 0x7ffd774f3d50 "", len = -164192016}, sip_params = {s = 0x0, len = -30810208}, headers = {s = 0x0, len = -181051830}, port_no = 8055, proto = 62988, type = 31908, flags = (unknown: 4128077712), transport = { s = 0x4774f37d0 <error: Cannot access memory at address 0x4774f37d0>, len = 2001680544}, ttl = {s = 0x669fa379 <error: Cannot access memory at address 0x669fa379>, len = -164192016}, user_param = { s = 0x5745ed7743b0 "'sip:+13194676789@us.test.com',NULL,0)", len = 1721781302}, maddr = {s = 0x400000000 <error: Cannot access memory at address 0x400000000>, len = -167513741}, method = { s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, lr = {s = 0x7ffd774f3810 "\340\070Ow\375\177", len = -167567737}, r2 = { s = 0x8000000 <error: Cannot access memory at address 0x8000000>, len = -343946861}, gr = {s = 0x7ffd774f3810 "\340\070Ow\375\177", len = -166977673}, transport_val = {s = 0x7ca4f60431d0 <sql_str> "@@w\355EW", len = -164192016}, ttl_val = {s = 0x7ffd774f3a90 "\330\267\027\366\244|", len = 23}, user_param_val = { s = 0x5745ed77417a "3071,3795,'6e1a9c34-cec5-4ebb-bcd9-3356d80bcf9e',' sip:+13194004696@sip.telnyx.com','c5SS3F27B8rDN',' sip:+13194676789@us.test.com','f9b9b2c48ef41ef90eb05056b0a794','tls: 10.16.0.74:520"..., len = -164192016}, maddr_val = {s = 0x7ca4f60dcf18 "`", len = -31060352}, method_val = {s = 0x0, len = -180627954}, lr_val = {s = 0x7ca4f60c1f77 <db_mysql_submit_query> "UH\211\345AWAVAUATSH\201\354\210", len = -166889584}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}} u = 0x7ca4f5817bc8 port = 0 dst_host = 0x7ca474b9751d i = -344941541 flags = 0 avp = 0x7ffd774f3c40 st = {flags = 3984015424, id = 22341, name = {n = -310010607, s = {s = 0x5745ed859d11 "", len = 2001679968}, re = 0x5745ed859d11}, avp = 0x66a04c36} sct = 0x60f60e6120 sjt = 0x5f60e6a50 rve = 0x8000000 mct = 0x0 rv = 0xffffffff rv1 = 0x5745eb7fc993 c1 = {cache_type = 10, val_type = RV_NONE, c = {avp_val = {n = 0, s = {s = 0x0, len = 0}, re = 0x0}, pval = {rs = {s = 0x0, len = 0}, ri = 2001679912, flags = 32765}}, i2s = "`6Ow\375\177\000\000\031\241\a\366\244|\000\000\250\003\000\000\000"} s = {s = 0xffffffff <error: Cannot access memory at address 0xffffffff>, len = -164179120} srevp = {0x3, 0x64870454268b4200} evp = {data = 0x0, obuf = {s = 0x0, len = 0}, rcv = 0x0, dst = 0x0, req = 0x0, rpl = 0x0, rplcode = 0, mode = 0} mod_f_params = {{type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = { number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, {type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}, { type = NOSUBTYPE, u = {number = 0, string = 0x0, str = {s = 0x0, len = 0}, data = 0x0, attr = 0x0, select = 0x0}}} __func__ = "do_action" #6 0x00005745eb40b35c in run_actions (h=0x7ffd774f3dc0, a=0x7ca4f61f0468, msg=0x7ca4f641a9f8) at core/action.c:1584 t = 0x7ca4f61f1a50 ret = 1 tvb = {tv_sec = 0, tv_usec = 0} tve = {tv_sec = 0, tv_usec = 0} tz = {tz_minuteswest = 60, tz_dsttime = 0} tdiff = 4131498432 __func__ = "run_actions" #7 0x00007ca474b92f8d in timer_handler (ticks=96802941, tl=0x7ca47b2f7950, data=0x7ca47b2f7868) at timer.c:216 msg = 0x7ca4f641a9f8 a = 0x7ca47b2f7868 ra_ctx = {rec_lev = 1, run_flags = 0, last_retcode = 1, jmp_env = {{__jmpbuf = {95957815091336, 8333654393230933038, 0, 2147483648, 134217728, 95957815380371, 8333654393012829230, 2510245087605373998}, __mask_was_saved = 0, __saved_mask = {__val = {137047930071236, 0, 25985340079663656, 140726605135616, 137045787154760, 70721158768, 137045787160648, 140726605135472, 137045776334895, 1095350878208, 137045782276504, 140735195070144, 140726605135504, 137045776144708, 0, 140726605135552}}}}} keng = 0x0 evname = {s = 0x7ca474b9715a "timer", len = 5} __func__ = "timer_handler" --Type <RET> for more, q to quit, c to continue without paging-- #8 0x00005745eb49e3c4 in slow_timer_main () at core/timer.c:1105 n = 12 ret = 4294967295 tl = 0x7ca47b2f7950 i = 1010 __func__ = "slow_timer_main" #9 0x00005745eb3a85bd in main_loop () at main.c:1765 i = 16 pid = 0 si = 0x0 si_desc = "udp receiver child=15 sock=10.16.0.74:5101 ( 35.202.133.184:5101)\000AOw\375\177\000\000d\366\023\376\244|\000\000 \000\000\000\060\000\000\000@AOw\375\177\000\000\200@Ow \375\177\000\000\000B\213&T\004\207d\223\311\177\353EW\000\000\240\274\065\366\244|\000" nrprocs = 16 woneinit = 1 __func__ = "main_loop" #10 0x00005745eb3b34c3 in main (argc=8, argv=0x7ffd774f4688) at main.c:2942 cfg_stream = 0x5745ed74d2a0 c = -1 r = 0 tmp = 0x7ffd774f5d9f "" tmp_len = 0 port = 0 proto = 0 ahost = 0x0 aport = 0 options = 0x5745eb7b8d78 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 1555962608 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 2 n_lst = 0x7ffd77580248 p = 0x7ca4fe189995 <handle_intel+197> "H\205\300uf\213t$\004M\215\216\244" st = {st_dev = 0, st_ino = 0, st_nlink = 0, st_mode = 0, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 0, st_blksize = 0, st_blocks = 0, st_atim = {tv_sec = 0, tv_nsec = 0}, st_mtim = {tv_sec = 0, tv_nsec = 0}, st_ctim = {tv_sec = 0, tv_nsec = 0}, __glibc_reserved = {0, 0, 0}} tbuf = "\b\000\000\000\000\000\000\000\210FOw\375\177\000\000\320FOw\375\177\000\000\330+,\376\244|\000\000\000\000\000\000\000\000\000\000\020\vH\376\244|\000\000\001", '\000' <repeats 19 times>, " \000\000\000(&G\376\244|\000\000\320EOw\375\177\000\000\312tH\376\244|\000\000\000\000\000\000\000\000\000\000@e*\376\244|\000\000\320\263+\376\244|\000\000\364EOw\375\177\000\000\006", '\000' <repeats 15 times>, "/", '\000' <repeats 39 times>, "\200\037\000\000\377\377", '\000' <repeats 186 times>... option_index = 0 long_options = {{name = 0x5745eb7baf46 "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x5745eb7b6524 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x5745eb7baf4b "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x5745eb7baf51 "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x5745eb7baf57 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x5745eb7baf60 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x5745eb7baf6a "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x5745eb7baf74 "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x5745eb7baf7f "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x5745eb7baf88 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x5745eb7baf93 "debug", has_arg = 1, flag = 0x0, val = 1032}, { name = 0x5745eb7baf99 "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __func__ = "main" (gdb) list 411 } 412 } while(RES_ROW_N(res) > 0); 413 414 for (i = 0; i < rd->carrier_num; ++i) { 415 for (j = 0; j < rd->carriers[i]->domain_num; ++j) { 416 if (rd->carriers[i]->domains[j]->sum_prob == 0.0) { 417 LM_ERR("All routes with carrier id %d (%.*s) and domain id %d (%.*s) have probability 0.\n", 418 rd->carriers[i]->id, rd->carriers[i]->name->len, rd->carriers[i]->name->s, 419 rd->carriers[i]->domains[j]->id, rd->carriers[i]->domains[j]->name->len, rd->carriers[i]->domains[j]->name->s); 420 } Please help to understand the reason for the crash and how to fix the problem. Thank you in advance. Regards *Maharaja Azhagiah*

3 7

Limiting Video calls(Volte to video conversion)
by dushy 28 Aug '24

28 Aug '24

Hello All, I am trying to limiting the number of video calls (converting Volte to Vilte). To reduce traffic on the enodeb. Its like if there are 10 volte calls going on then I should be able to limit the number calls that can be converted to Vilte. I tried to write a route in kamailio_pcscf.cfg using module ims_dialog. But it doesnt work and get some errors while doing a modparam. Tried another method of counting eRAB at eNB (ENBUEID and ERABID pair with qci-2). This worked, but after sometime the extra call got established. Looks like traffic is going through qci 9 bearer. Is there any method where we can achieve this. Thanks in Advance. Regards, Dushyanth

1 0

Re: certificate verify failed (sni: unknown) integration with ms teams
by Muhammad Sohaib 27 Aug '24

27 Aug '24

Hi Batanai, After disabling #!define WITH_NAT in Kamailio, calls from the Asterisk endpoint to Microsoft Teams are functioning correctly. Both inbound and outbound calls exhibit proper two-way audio and no disconnection issues but there is no Bye when hangup from ms teams. However, I am encountering a problem with outbound calls originating from Microsoft Teams. When a call is placed from MS Teams, the recipient's phone rings as expected. Once the call is answered, Microsoft Teams disconnects the call immediately. The root of the issue seems to be that the ACK message is not being sent or received properly. As a result, Microsoft Teams does not receive the ACK, leading to an immediate call termination. Additionally, Asterisk does not receive a BYE message from Kamailio. *Summary of the Issue:* - *Inbound Calls*: From Asterisk to MS Teams work fine, with full audio and no disconnections but no BYE from ms teams. - *Outbound Calls*: From MS Teams to Asterisk result in the call being disconnected immediately after being answered due to a missing ACK, and Asterisk does not receive a BYE message. Thanks, On Tue, Aug 27, 2024 at 5:45 PM Batanai Musiiwa <batanai.musiiwa(a)etel.co.zw> wrote: > Hi @ > Muhammad. May you advise if the calls are disconnecting in either > direction? > > ; > > On Tue, Aug 27, 2024 at 11:05 AM Muhammad Sohaib via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> Hi everyone, >> >> Thank you all for your kind help with the TLS configuration. Palany's >> suggestion worked flawlessly, and now both inbound and outbound calls are >> functioning. >> >> However, I'm facing a one-way audio issue, and calls are disconnected >> after 30 seconds. I suspect this might be related to NAT. Here’s the >> current setup: >> >> *Asterisk <===> Kamailio (as SBC) <=====> MS Teams* >> >> Could this be related to my NAT configuration, or is there something else >> I should investigate? Any guidance would be greatly appreciated. >> >> Thanks again for your support! >> >> On Thu, Aug 15, 2024 at 11:02 PM palany <palany(a)advancedzim.com> wrote: >> >>> >>> >>> Hi Muhammad >>> >>> >>> >>> Can you try your tls cfg as below and make sure your certificates have >>> the right permissions. >>> >>> >>> >>> [server:default] >>> >>> method = TLSv1.2+ >>> >>> verify_certificate = no >>> >>> require_certificate = no >>> >>> private_key = /etc/letsencrypt/live/MYDOMAIN/privkey.pem >>> >>> certificate = /etc/letsencypt/live/MYDOMAIN/fullchain.pem >>> >>> >>> >>> [client:default] >>> >>> method = TLSv1.2+ >>> >>> verify_certificate = no >>> >>> require_certificate = no >>> >>> private_key = /etc/letsencrypt/live/MYDOMAIN/privkey.pem >>> >>> certificate = /etc/letsencrypt/live/MYDOMAIN/fullchain.pem >>> >>> *From:* Muhammad Sohaib via sr-users [mailto:sr-users@lists.kamailio.org] >>> >>> *Sent:* Thursday, 15 August 2024 3:46 PM >>> *To:* Kamailio (SER) - Users Mailing List >>> *Cc:* Muhammad Sohaib >>> *Subject:* [SR-Users] certificate verify failed (sni: unknown) >>> integration with ms teams >>> >>> >>> >>> Dear all, >>> >>> >>> >>> Trying to integrate Kamailio with MS Teams by following >>> https://skalatan.de/en/blog/kamailio-sbc-teams >>> >>> >>> >>> kamcmd dispatcher.list | egrep "URI|FLAGS" >>> >>> >>> URI: sip: >>> sip.pstnhub.microsoft.com;transport=tls >>> FLAGS: IP >>> URI: sip: >>> sip2.pstnhub.microsoft.com;transport=tls >>> FLAGS: IP >>> URI: sip: >>> sip3.pstnhub.microsoft.com;transport=tls >>> FLAGS: IP >>> >>> >>> >>> Kamailio Logs: >>> >>> /usr/local/sbin/kamailio[412158]: INFO: <script>: Sent out tm request: >>> OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0#015#012 >>> Via: SIP/2.0/TLS >>> x.x.x.x:5061;branch=z9hG4bK5dad.92de50b2000000000000000000000000.0#015#012 >>> To: <sip:sip.pstnhub.microsoft.com;transport=tls>#015#012 >>> From: <sip:test.mytest.com >>> >;tag=5d0939b82abe9b1bbf185d963b6e6c88-edeb3c71#015#012 >>> CSeq: 10 OPTIONS#015#012 >>> Call-ID: 2db6bede5631d9b6-412158(a)88.99.244.106#015#012 >>> <http://2db6bede5631d9b6-412158@88.99.244.106#015%23012> >>> Max-Forwards: 70#015#012Content-Length: 0#015#012 >>> User-Agent: kamailio (5.8.2 (x86_64/linux)) >>> >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1312]: >>> tls_h_read_f(): protocol level error >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_util.h:49]: >>> tls_err_ret(): TLS write:error:0A000086:SSL routines::certificate verify >>> failed (sni: unknown) >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1316]: >>> tls_h_read_f(): src addr: 52.114.75.24:5061 >>> /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1319]: >>> tls_h_read_f(): dst addr: x.x.x.x:0 >>> /usr/local/sbin/kamailio[412165]: ERROR: <core> [core/tcp_read.c:1524]: >>> tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fa74d265d40 r: >>> 0x7fa74d265e68 (-1) >>> >>> >>> tls.cfg: >>> >>> [server:default] >>> method = TLSv1.2+ >>> verify_certificate = yes >>> require_certificate = yes >>> private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem >>> certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> server_name = test.mytest.com >>> >>> [client:default] >>> method = TLSv1.2+ >>> verify_certificate = yes >>> require_certificate = yes >>> private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem >>> certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem >>> >>> Please suggest what I am missing. >>> >>> --- >>> >>> Thanks, >>> >> >> >> -- >> Thanks, >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > -- Thanks,

1 0

certificate verify failed (sni: unknown) integration with ms teams
by Muhammad Sohaib 27 Aug '24

27 Aug '24

Dear all, Trying to integrate Kamailio with MS Teams by following https://skalatan.de/en/blog/kamailio-sbc-teams kamcmd dispatcher.list | egrep "URI|FLAGS" URI: sip:sip.pstnhub.microsoft.com ;transport=tls FLAGS: IP URI: sip:sip2.pstnhub.microsoft.com ;transport=tls FLAGS: IP URI: sip:sip3.pstnhub.microsoft.com ;transport=tls FLAGS: IP Kamailio Logs: /usr/local/sbin/kamailio[412158]: INFO: <script>: Sent out tm request: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0#015#012 Via: SIP/2.0/TLS x.x.x.x:5061;branch=z9hG4bK5dad.92de50b2000000000000000000000000.0#015#012 To: <sip:sip.pstnhub.microsoft.com;transport=tls>#015#012 From: <sip:test.mytest.com >;tag=5d0939b82abe9b1bbf185d963b6e6c88-edeb3c71#015#012 CSeq: 10 OPTIONS#015#012 Call-ID: 2db6bede5631d9b6-412158(a)88.99.244.106#015#012 Max-Forwards: 70#015#012Content-Length: 0#015#012 User-Agent: kamailio (5.8.2 (x86_64/linux)) /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1312]: tls_h_read_f(): protocol level error /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_util.h:49]: tls_err_ret(): TLS write:error:0A000086:SSL routines::certificate verify failed (sni: unknown) /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1316]: tls_h_read_f(): src addr: 52.114.75.24:5061 /usr/local/sbin/kamailio[412165]: ERROR: tls [tls_server.c:1319]: tls_h_read_f(): dst addr: x.x.x.x:0 /usr/local/sbin/kamailio[412165]: ERROR: <core> [core/tcp_read.c:1524]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fa74d265d40 r: 0x7fa74d265e68 (-1) tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem server_name = test.mytest.com [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/letsencrypt/live/test.mytest.com/privkey.pem certificate = /etc/letsencrypt/live/test.mytest.com/fullchain.pem ca_list = /etc/letsencrypt/live/test.mytest.com/fullchain.pem Please suggest what I am missing. --- Thanks,

5 5

I have mo media and I must park
by Alexandru Covalschi 27 Aug '24

27 Aug '24

Hello list! Please help me understand if putting my b2bua in sensory deprivation is actually the best approach to punish it for sole reason of its existence. The background of this horror story is pretty cliche, call center needs a way to infinitely expand, so I'm building a scaling telephony core. Our main protagonist is an innocent b2bua with a very grounded backstory: step 0: receive external instruction to start a call step 1: originate step 2: park step 3: play moh until: - - a) told to pass it to other component - - b) told to hang up - - c) hang up on local timeout A simplified diagram of the call flow looks something like this (repeating names are same components): ### Signaling ### Remote Destination ^ Kamailio-external ^ Freeswitch-originator V Freeswitch- queue V Kamailio-local V Agent ### end of signaling ### ### Media ### Remote Destination ^ Rtpengine-external ^ Freeswitch-originator V Freeswitch-queue V Rtpengine- local V Agent ### end of media ### Due to specifics of the setting, re-negotiating the media and any non-essential signaling with remote destination is a very much last resort. And, as you can imagine, after step3a (pass it to other component), the character development of Freeswitch-originator service stops completely. Well, at least it is doing an honest job by playing the music on hold, which a little bit justifies its presence on the media path. But not for long. In the world of near future, pull request number 3956 is merged with the master branch, exposing the rtpengine API and leaving me no reason not to move MoH to a kernel media player. At this moment I feel a wild urge to get rid of Freeswitch-originator at all, but then I read the sources for mohq and dialog to remind myself that Kamailio is a proxy and proxies can't do this... So I fill b2bua's dialplan with 387.44 million lines of hateful comments and remind it to rcvonly every time it tries to warn the others about the grim fate. When the queue designates an agent to his call, theoretically I don't even need the second freswitch as well - if agen't media session has been established (sometimes ICE take tooo long) with Rtpengine- local, I can bridge their media sessions using publish and subscribe. I'd like to ask your opinion, list: do you see any plotholes potential issues with such deisgn? I'd honestly like to set Freeswitch to bypass-media, so it wont' waste ports and cpu for nothing at all. Are there any issues if I add body for the iniitial invite in the request_route as if it was received in there? Maybe there's a better way, overall? Any critique is welcome! Thank you -- AC

1 0

How to invoke RTPengine for re-Invite 200 OK within dialog
by Shah Hussain Khattak 27 Aug '24

27 Aug '24

Hello Experts, I need your advice on the following issue if you can kindly help. I am working on a hold call flow: UA - > Kamailio (RTP Engine) - > FS initial call establishes fine, RTP engine is invoked and SDP media IP is modified correctly. Then 1. I am receiving a hold re-invite with a send-only 2. I see Kamailio is changing the media IP in the SDP (with my RTP engine IP) when it sends the Invite to my core FS server. 3. FS sends 200 OK with a=recvonly 4. Kamailio relays the message as it is to the other leg. can someone advise me, on how can I invoke rtpengine at step 4, so the SDP is sent with my media server IP in the 200 OK? --- 200 OK from FS ---- SIP/2.0 200 OK Via: SIP/2.0/UDP 13.54.01.02:5060;branch=z9hG4bKd2e8.8a573af8d927f67c1db4550676f8be88.0;i=1 Via: SIP/2.0/TCP 192.168.10.117:51280;received=218.33.01.02;rport=51280;branch=z9hG4bKPj7d1d8968830d4ed3922eec7da7cdc17c;alias Record-Route: <sip:13.54.01.02:5060;r2=on;lr=on> Record-Route: <sip:13.54.01.02;transport=tcp;r2=on;lr=on> From: <sip:0403225908@trunk.xyz.corp>;tag=6a180ffa7f824972a2c59ba47f293acb To: <sip:+61285038000@trunk.xyz.corp>;tag=DZrpcpgS1BDKF Call-ID: 4154062196634c34a2dbcb6b612bf44c CSeq: 12263 INVITE Contact: <sip:+61285038000@54.88.54.88:5060;transport=udp> User-Agent: UserAgentX/v1.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: timer, path, replaces Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1724635755 1724635757 IN IP4 54.88.54.88 s=FreeSWITCH c=IN IP4 54.88.54.88 t=0 0 m=audio 21202 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly a=ptime:20 a=rtcp:21203 IN IP4 54.88.54.88 --- 200 OK relayed towards the UA ---- 2024/08/26 17:22:39.923639 172.16.8.126:5060 -> 218.33.78.33:51280 SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.10.117:51280;received=218.33.01.02;rport=51280;branch=z9hG4bKPj7d1d8968830d4ed3922eec7da7cdc17c;alias Record-Route: <sip:13.54.01.02:5060;r2=on;lr=on> Record-Route: <sip:13.54.01.02;transport=tcp;r2=on;lr=on> From: <sip:0403225908@trunk.xyz.corp>;tag=6a180ffa7f824972a2c59ba47f293acb To: <sip:+61285038000@trunk.xyz.corp>;tag=DZrpcpgS1BDKF Call-ID: 4154062196634c34a2dbcb6b612bf44c CSeq: 12263 INVITE Contact: <sip:+61285038000@54.88.54.88:5060;transport=udp> User-Agent: UserAgentX/v1.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: timer, path, replaces Session-Expires: 1800;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 266 v=0 o=FreeSWITCH 1724635755 1724635757 IN IP4 54.88.54.88 s=FreeSWITCH c=IN IP4 54.88.54.88 t=0 0 m=audio 21202 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=recvonly a=ptime:20 a=rtcp:21203 IN IP4 54.88.54.88 my configs for within dialog route: ##### route[WITHINDLG] { append_hf("X-Test-Header: route_WITHINDLG\r\n"); if (has_totag()) { if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if (is_method("ACK")) { xlog("L_INFO", "Relaying ACK\n"); } if (is_method("INVITE|UPDATE")) { if (has_body("application/sdp") && search_body("m=image") && search_body("T38")) { xlog("L_INFO", "T.38 re-INVITE detected, skipping rtpengine invocation\n"); } else { xlog("L_INFO", "Non-T.38 INVITE detected, invoking rtpengine\n"); rtpengine_manage("replace-origin replace-session-connection"); } } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { route(PRESENCE); exit; } if (is_method("ACK")) { if (t_check_trans()) { t_relay(); exit; } else { exit; } } sl_send_reply("404","Not here"); } exit; } } ##### looking forward to your usual guidance. Thank you! Regards, Shah

2 2

socket names for command line parameter "listen"
by Xenofon Karamanos 23 Aug '24

23 Aug '24

Hello all, I wanted to inform you about an upcoming change regarding the syntax we use for defining socket names in the listen (-l) command line argument. Current Syntax: -l udp:10.10.10.10:5060/11.11.11.11:5060 Proposed Options: 1. -l udp:10.10.10.10:5060/11.11.11.11:5060=s1 2. -l s1=udp:10.10.10.10:5060/11.11.11.11:5060 ( A draft PR about this option can be found in https://github.com/kamailio/kamailio/pull/3954) [https://opengraph.githubassets.com/69f8a4a1d2c8988372b4942f84bde95de94502d2…]<https://github.com/kamailio/kamailio/pull/3954> main: Add support for naming socket in cli by xkaraman · Pull Request #3954 · kamailio/kamailio<https://github.com/kamailio/kamailio/pull/3954> Pre-Submission Checklist Commit message has the format required by CONTRIBUTING guide Commits are split per component (core, individual modules, libs, utils, ...) Each component has a single... github.com Please note that the socket name (s1) in both options is/should be optional. The change will allow for better clarity and flexibility when defining sockets, particularly when a name needs to be assigned and match the config file style. I would appreciate your feedback on which syntax you find more intuitive or any concerns you might have with these options. Your input is important to ensure a smooth transition. Best regards, Xenofon

2 2

Kamailio cluster options
by Ali Taher 22 Aug '24

22 Aug '24

Hello all, I'm using Kamailio as SIP redirect (act as routing decision for SBC) depending mainly on sqlops and db_postgres modules, and it's working perfectly. What I need now is to have a kind of cluster of Kamailio servers for high availability purpose. What is the best option for me ? BR

2 1

push server tcp problem
by Bernd Krueger-Knauber 20 Aug '24

20 Aug '24

Hi, I implemented an apple and android push server/proxy with kamailio 5.8.1. The server/proxy is transparent. It works fine with UDP, since the registration after the push comes from the same ip and port. So t_continue works without problems. (We use the new registration as signal thet the phone is ready) If we use tcp at the client, he gets the push, wakes up and send the new registration on an other port. The SIP stack uses now an other connection. t_continue fails, because it uses the old data of the original invite acccording to the first registration. Is it possible to change the stored connection values according to the new registration data? Best regards, Bernd

2 2

low performance with udp_children and async_workers
by elhar.mohamed＠gmail.com 19 Aug '24

19 Aug '24

Hello, Based on business requirements, 35% of invites should be handled by async_workers and 65% with be handled by main children. The target is 1500cps. I'm using the following conf : children=64 async_worker=128 when doing performance tests using SIPp, I figured out the 1500cps is well supported by the conf above, but when I updated the data in database to not use anymore async workers, the cps is getting decreased and it was between 200 and 500. This a client requirements is to be able to enable/disable feature that needs to be handled by async workers. Could you please help me on this? Thanks in advance

4 12

Inbound call to webRTC failing with TCP/TLS connection (id: 0) for WebSocket could not be found
by hollis.rhonda＠gmail.com 16 Aug '24

16 Aug '24

We're seeing an issue where some webRTC clients are not receiving inbound calls. Kamailio logs show the following error: WARNING: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: <core> [core/msg_translator.c:3007]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found ERROR: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: <core> [core/msg_translator.c:2086]: build_req_buf_from_sip_req(): could not create Via header ERROR: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_fwd.c:484]: prepare_new_uac(): could not build request ERROR: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_fwd.c:1764]: t_forward_nonack(): failure to add branches DEBUG: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_funcs.c:358]: t_relay_to(): t_forward_nonack returned error -2 (-2) DEBUG: {1 609669922 INVITE BW152025538120824-895739618(a)10.103.43.11}: tm [t_funcs.c:376]: t_relay_to(): -2 error reply generation delayed It seems to be happening after the client has been connected to Kamailio for more than 24 hours. The socket connection details look correct in the external DB and in the output of ws.dump, core.tcp_list, and ul.dump. A restart of Kamailio or a reload of the webRTC client, triggering a new websocket connection, will clear the issue. However, we haven't been able to determine exactly when and why a client ends up in this state. While a client is in this state, REGISTER requests are still handled successfully, they can make outbound calls and TCP keepalives from the client are working. I'm still working on trying to debug this with GDB so I can see what the value of send_info is in https://github.com/kamailio/kamailio/blob/master/src/core/msg_translator.c#…, but I wasn't having luck with multiple child processes running, and now my client has reloaded and is working again. I've restarted Kamailio with a single child process, so when it does start failing again, I may have better luck with GDB. Any thoughts on what might be causing clients to get into this state? Whatever additional information I can provide that might help, I'm happy to share. Thanks!

2 4

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 16 Aug '24

16 Aug '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

3 2

[http_client] KEMI use "curl_connect_post" send http request error
by 17603019528＠163.com 16 Aug '24

16 Aug '24

ALL: Hi, here is my lua script: ``` local body = '{"ip":"172.16.4.111", "ports":[5261, 5260]}' local resp = "" local code = KSR.http_client.curl_connect_post("configapi", "", "application/json", body, resp) ``` the error is : ``` 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[0] for: curl_connect_post is str: configapi 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[1] for: curl_connect_post is str: 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[2] for: curl_connect_post is str: application/json 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[3] for: curl_connect_post is str: {"ip":"172.16.4.111", "ports":[5261, 5260]} 24(2406) DEBUG: app_lua [app_lua_api.c:1024]: sr_kemi_lua_exec_func_ex(): param[4] for: curl_connect_post is str: 24(2406) ERROR: <core> [core/pvapi.c:429]: pv_cache_get(): invalid parameters 24(2406) ERROR: http_client [http_client.c:739]: ki_curl_connect_post(): failed to get pv spec for: ``` I know the error means: "resp" type is not right, what is the correct type ? I search google for this issue, no one knows, someone please help me. Thinks

2 2

TLS module doesn't support TLSV1
by o.atef＠fiberme.com 15 Aug '24

15 Aug '24

Hello Brothers, I've installed kamailio throw "apt install" on Debian and it's installed version: kamailio 5.6.3 (x86_64/linux). Now I've a big problem that kamailio cannot running with TLSv1 and it has to be TLSv1.2+ as tls.cfg doc said: # We do not enable anything else than TLSv1.2+ # over the public internet. Clients do not have # to present client certificates by default. How could I avoid this restriction please to enable TLSv1? Thank you,

5 5

[PHONENUM] Match with country on Lua/KSR
by João Silva 14 Aug '24

14 Aug '24

Hello everyone! The function phonenum_match_cn from module PHONENUM is not available on KSR. Is there any work around to use it on Lua config? Best regards, Joao

2 2

Incoming call with one one way audio and ctting off after thirty seconds
by palany 14 Aug '24

14 Aug '24

I have integrated Microsoft teams and freepbx using Kamailio version 5.8 as my SBC. Calls from freepbx to teams are working fine with audio and I am having challenges with calls from teams to freepbx. The calls rings and after answering there is only way audio and the call drops off after thirty seconds. My pbx is behind a microtik router and I have port forwarded my sip port and rtp ports to freepbx and have also disabled sip alg. My Kamailio is hosted in AWS. I have attached both my pcap from Kamailio, freepbx and my Kamailio.cfg. I am thinking It's a misconfiguration challenge or something else. Please help I am failing to see my challenge. Best Regards,

3 3

Authentication
by Vishal Pai 13 Aug '24

13 Aug '24

Hello All I am using kamailio sip account as gateway at Freeswitch side but when I enable caller id from = true then call from Freeswitch to kamailio didn’t authenticate As I check it’s about from header when I disable caller id from = false the call get authenticate since it get sip username and domain but when there is caller id the from header get callerid@ip address Can anyone guide me Thanks

2 1

Kamailio hangs on get_profile_size method.
by satyaprakash ch 12 Aug '24

12 Aug '24

HI, We are facing a hang issue with Kamailio for the get_profile_size method. Randomly, the Kamailio got hangs, and the CPU went 100%. During the hanging time, we generated the TRAP and we could see that the hang happened on the get_profile_size method. We are getting this hang randomly in dlg_profile.c of the following lines: The hang occurs, causing the CPU to spike to 100% at one of the following lines randomly: 1. if(value->len == ph->value.len && memcmp(value->s, ph->value.s, value->len) == 0) { 2. ph = ph->next; 3. while(ph != profile->entries[i].first); During our analysis, we found that PROCESS ID 7608 gets a spike of 100% utilization. We suspect there may be an issue with the processing of the 'ph' variable, but we couldn't identify the root cause. I've attached the Kamailio TRAP for further details. Could you please suggest what might be causing this problem and how we can resolve it? Thank you.

1 1

Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 10 Aug '24

10 Aug '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria [aws](https://www.igmguru.com/cloud-computing/aws-developer-certification-tr…

4 4

Kamailio hangs on get_profile_size method.
by satyaprakash ch 09 Aug '24

09 Aug '24

HI, We are facing a hang issue with Kamailio for the get_profile_size method. Ramdomly, the Kamailio got hangs, and the CPU went 100%. During the hanging time, we generated the TRAP and we could see that the hang happened on the get_profile_size method. We are getting this hang randomly in dlg_profile.c of the following lines: The hang occurs, causing the CPU to spike to 100% at one of the following lines randomly: 1. if(value->len == ph->value.len && memcmp(value->s, ph->value.s, value->len) == 0) { 2. ph = ph->next; 3. while(ph != profile->entries[i].first); During our analysis, we found that PROCESS ID 7608 gets a spike of 100% utilization. We suspect there may be an issue with the processing of the 'ph' variable, but we couldn't identify the root cause. I've attached the Kamailio TRAP for further details. Could you please suggest what might be causing this problem and how we can resolve it? Thank you.

1 0

Incoming call with one one way audio and cutting off after thirty seconds
by palany 08 Aug '24

08 Aug '24

I have integrated Microsoft teams and freepbx using Kamailio version 5.8 as my SBC. Calls from freepbx to teams are working fine with audio and I am having challenges with calls from teams to freepbx. The calls rings and after answering there is only way audio and the call drops off after thirty seconds. My pbx is behind a microtik router and I have port forwarded my sip port and rtp ports to freepbx and have also disabled sip alg. My Kamailio is hosted in AWS. Below are google drive links logs for the incoming call in Kamailio , freepbx and my Kamailio.cfg. I am thinking It's a misconfiguration challenge or something else. Please help I am failing to see my challenge. kamailio.pcap - https://drive.google.com/file/d/1aoJhksgwcH-52unwxuyzn8GAmG9PEnOM/view?usp=s haring freepbx.pcap - https://drive.google.com/file/d/1JJXs9UOUFak5Y4uWI_ARB56vQfVYfSDj/view?usp=s haring kamailio.cfg - https://drive.google.com/file/d/1v2ZtylOf8fIMYH4OckGmnP489PO7WI7J/view?usp=s haring Best Regards

2 2

Dialog timer reset on UPDATE
by Alex Balashov 08 Aug '24

08 Aug '24

Hi, I have a dialog where I have set: dlg_set_property("timeout-noreset"); But it seems the timeout is reset anyway when handling an in-dialog UPDATE request, at least, on Kamailio 5.5.4. This does not seem to happen with reinvites. I haven't been able to get more information thus far, other than to be able to say that there is nothing unusual about the UPDATE request and there are no additional errors at log level 2. I was wondering if anyone else has encountered this. -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

1 0

app_python3 method not found or is not callable
by Westhuis, Johannes 08 Aug '24

08 Aug '24

Hi, we are using kamailio with app_python3. It works very good, but in some cases, there are error logs stating: app_python3 [python_exec.c:91]: apy_exec(): <python method> not found or is not callable It happens sporadically, and a restart of kamailio fixes it. As far as i can see it might be linked to concurrent arrival of several messages. Best regards, Johannes # kamailio -v version: kamailio 5.5.6 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: compiled on 12:01:03 Jul 22 2024 with gcc 10.2.1

1 0

http_async_query on carrierfailureroute changes rd back to carrieroute primary
by Maharaja Azhagiah 07 Aug '24

07 Aug '24

Hi I am running kamailio 5.4 on debian I have carrierfailureroute configured incase of primary service provider fails. I also have Stirshaken configured to add Identity header on outbound calls. Issue is when call fail overs to carrierfailureroute, http_async_query changes $ru to the primary carrier From the debug logs, when primary carrier sends a 488 (primary carrier expects SIP TLS but my call is UDP - to test the failover scenario) 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} tmx [t_var.c:561]: pv_get_tm_reply_code(): reply code is <488> 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:178]: set_next_domain_on_rule(): searching for matching routing rules39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:197]: set_next_domain_on_rule(): next_domain is 47987 Carrier route rewrites the failover carrier 39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} carrierroute [cr_func.c:706]: cr_do_route(): uri 14371234567 was rewritten to sip:14371234567@sip.primaryprovider.com, carrier 1, domain 47987 Before http_async_query rd and ru are still the failover carrier 39(285) INFO: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} <script>: [callid: 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn] - [cfg:2976] - Debug testing ----- rd is sip.primaryprovider.com ----- ru is sip:14371234567@sip.primaryprovider.com 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} http_async_client [async_http.c:469]: async_send_query(): transaction suspended [5261:1830449764] 39(285) DEBUG: {1 18398 INVITE 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn} http_async_client [async_http.c:625]: async_push_query(): query sent [ https://authn-uat.ccid.neustar.biz/ccid/authn/v2/identity?apiKey=randomkey] (0x7fdcad097e60) to worker 1 However, when the route is being called after the http_async_query it changes to the primary one: 26(272) DEBUG: tm [t_lookup.c:1612]: t_lookup_ident_filter(): transaction found 26(272) DEBUG: http_async_client [async_http.c:235]: async_http_cb(): resuming transaction (5261:1830449764) 26(272) DEBUG: tm [t_lookup.c:1612]: t_lookup_ident_filter(): transaction found 26(272) INFO: <script>: [callid: 8EmmsLqNuMRYBduMqFgX3w4JHAn4C2xn] - [cfg:2995] - Debug testing ----- rd is 1.2.3.4 ----- ru is sip:14371234567@1.2.3.4:5061;transport=TLS Due to this, call keeps going to the primary and it fails if ( http_async_query(STIRSHAKEN_AS_URL, "AS_RESPONSE") == -1 ) { xlog("L_ERR ", "[cfg:$cfg(line)] Failed to connect AS service for token $fu -> $tu \n"); return; } route[AS_RESPONSE] { xlog("L_INFO", "[callid: $ci] - [cfg:$cfg(line)] - Debug testing ----- rd is $rd ----- ru is $ru\n"); if ($http_ok) { xlog("L_INFO", "[cfg:$cfg(line)] Resuming outbound call transaction for $fu -> $tu Received - $http_rb \n"); # Add identity and Date headers if (jansson_get("identity", $http_rb, "$var(identity)")) { insert_hf("Identity: $var(identity)\n", "Content-Length"); } if (jansson_get("date", $http_rb, "$var(date)")) { if ($hdr(Date) != $null){ remove_hf("Date"); } insert_hf("Date: $var(date)\n", "Identity"); } } else { xlog("L_ERR", "[cfg:$cfg(line)] Resuming outbound call transaction. Error - $http_err)\n"); } route(RELAY); exit; } Please help to understand why rd / ru changes to primary carrier. Regards, Maharaja Azhagiah

3 7

Kamailio hang Issue
by satyaprakash ch 06 Aug '24

06 Aug '24

Hi, We are encountering a hang issue after deploying the following code: *route(DLG_MANAGE);* *if ($dlg_var(voip_id) == $null) { $dlg_var(voip_id) = $hdr(X-Voip-ID);}route[DLG_MANAGE] { if (is_method("INVITE")) { dlg_manage(); }}* This code includes the set_dialog_profile. We have this setup with 2 proxy servers with DMQ enabled, but the issue only occurs on one proxy. When the Kamailio proxy hangs, no logs are written. If you have any suggestions to resolve this issue, please help. Thanks.

2 2

Kamailio-IMS: After a abornamal Call, other call is not working (even after a new registration)
by deep.upreti＠gmail.com 05 Aug '24

05 Aug '24

Hello, Can anyone help me to resolve one issue? scenario is - 1- Register two UE (UE-A and UE-B) 2- Call from UE-A to UE-B for a specific duration (lets say: 100 seconds) 3 Verify if call is working: Answer is YES, call is working. 4. shutdown the eNB 5. Observation: No BYE/De-registration message is not coming as expected. 6. Again start eNB and register same UE (UE-A and UE-B) 7. Register UE-A and UE-B both 8. It will not remove previous contacts or not updating the reg-state. 9. I am running with Rx enable mode.

1 1

Multiple TLS connections to the same IP:Port
by Nick Digalakis 05 Aug '24

05 Aug '24

Hello everyone, I am trying to use a single Kamailio server to register to multiple MS Teams Direct Routing endpoints. The config snippet I am using is this: sht_iterator_start("i1", "teams-endpoints"); while(sht_iterator_next("i1")) { $var(teams_endpoint) = $shtitkey(i1); $xavp(tls=>server_name) = $var(teams_endpoint); $xavp(tls[0]=>server_id) = $var(teams_endpoint); $uac_req(method)="OPTIONS"; $uac_req(ruri)="sip:sip.pstnhub.microsoft.com:5061;transport=tls"; $uac_req(furi)="sip:" + $var(teams_endpoint); $uac_req(turi)="sip:sip.pstnhub.microsoft.com:5061;transport=tls"; $uac_req(hdrs)="Contact: <sip:" + $var(teams_endpoint) + ":" + "5061" + ";transport=tls>\r\n"; ### Create a unique Call-ID based on the Timestamp and the Message Body in MD5 $var(unhashed_cid) = $TV(Sn) + $mb + ""; $uac_req(callid)=$(var(unhashed_cid){s.md5}); uac_req_send(); } sht_iterator_end("i1"); When the HTable has only one endpoint, everything works fine. When I add a second endpoint, the first one continues to work but the second one failing with the error from Microsoft: Q.850;cause=63;text="85babcde-e0b5-4a85-8f4a-12345678c9ae;SBC certificate is not issued correctly. Provided trunk FQDN 'endpoint-02.domain.com' is not included in certificate's CN or SAN list. Certificate allows following FQDNs only: endpoint-01.domain.com. After some digging around, I realized that all endpoint after the first fail because Kamailio is re-using the same TLS connection for all subsequent OPTIONS as well, but of course the connection has been established with the certificate of the first endpoint. I have tested the certificates by switching around the first endpoint, so that shouldn't be a problem. Is there any way I can force it to establish a new TCP/TLS connection for each subsequent request? Any help would be much appreciated, I have been pulling my hair out with this one! Best regards, Nick [https://s-install.avcdn.net/ipm/preview/icons/icon-envelope-tick-green-avg-…]<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…> Virus-free.www.avg.com<http://www.avg.com/email-signature?utm_medium=email&utm_source=link&utm_cam…>

2 2

How to log datetime in files
by 陈子妍 04 Aug '24

04 Aug '24

Hello! I am working on with Kamailio, I have already understood to add log prefix like `log_prefix="NAT {$rm $mt $hdr(CSeq) $ci}: "`, but I also want to record event time, so how can I config `datetime` in log files. Thanks~

3 2

Why does $T(id_index) return 0 even though t_suspend() is executed successfully?
by mm e 01 Aug '24

01 Aug '24

if(t_suspend()) { $dlg_var(start_time) = $timef(%Y/%m/%d %H:%M:%S); xlog("L_NOTICE", "==> processing suspended in transaction [$T(id_index):$T(id_label)] "); } else { xlog("L_ERR", "t_suspend err"); exit; } Sometimes, I get this log==> processing suspended in transaction [0:13614] "，as far as I know, 0 means that the value cannot be obtained. Executing t_continue(tindex, tlabel, rtname) with parameter 0 will fail.by the way, It's a question of probability.

1 0

Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 31 Jul '24

31 Jul '24

Hello Daniel, Regarding your question, I think only the avp(msg) should be destroyed as others avp's can be used elsewhere in the script. I tried the lastest git master and when restarting kamailio I got a segmentation fault: Jul 18 21:17:16 labcom070 systemd[1]: Started Kamailio - the Open Source SIP Server. Jul 18 21:17:17 labcom070 kernel: kamailio[2285]: segfault at 0 ip 00000000007eb66a sp 00007ffd7f80cc00 error 6 in kamailio[400000+533000] Jul 18 21:17:17 labcom070 kernel: Code: 41 56 41 55 41 54 53 48 81 ec c8 00 00 00 48 89 bd 48 ff ff ff 48 89 b5 40 ff ff ff 48 89 95 38 ff ff ff 48 8b 85 48 ff ff ff <48> c7 00 00 00 00 00 48 8b 85 48 ff ff ff c7 40 08 00 00 00 00 48 Jul 18 21:17:17 labcom 070 systemd[1]: Started Process Core Dump (PID 2286/UID 0). Jul 18 21:17:18 labcom070 systemd-coredump[2287]: Resource limits disable core dumping for process 2285 (kamailio). Jul 18 21:17:18 labcom070 systemd-coredump[2287]: Process 2285 (kamailio) of user 993 dumped core. Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Main process exited, code=dumped, status=11/SEGV Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Failed with result 'core-dump'. Jul 18 21:17:18 labcom070 systemd[1]: systemd-coredump(a)6-2286-0.service: Succeeded. Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Service RestartSec=100ms expired, scheduling restart. Jul 18 21:17:18 labcom070 systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 1. Jul 18 21:17:18 labcom070 systemd[1]: Stopped Kamailio - the Open Source SIP Server. Jul 18 21:17:18 labcom070 kernel: kamailio[2298]: segfault at 0 ip 00000000007eb66a sp 00007ffcd101c240 error 6 in kamailio[400000+533000] Jul 18 21:17:18 labcom070 kernel: Code: 41 56 41 55 41 54 53 48 81 ec c8 00 00 00 48 89 bd 48 ff ff ff 48 89 b5 40 ff ff ff 48 89 95 38 ff ff ff 48 8b 85 48 ff ff ff <48> c7 00 00 00 00 00 48 8b 85 48 ff ff ff c7 40 08 00 00 00 00 48 Regards, Vanderlei ________________________________ De: Daniel-Constantin Mierla <miconda(a)gmail.com> Enviado: quinta-feira, 18 de julho de 2024 02:51 Para: Vanderlei Torres Batistela <v.batistela(a)hotmail.com>; Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Assunto: Re: [SR-Users] Possible Kamailio memmory leak - version 5.8.2 Hello, indeed the previous commit was destroying a single avp, I pushed a new one to destroy all of them with that name. I am wondering if all avps, no matter the name should be destroyed. Anyhow, could you fetch the latest git master version and try your scenario again? Cheers, Daniel

1 0

[TM] [ASYNC] : Freeing transaction after sending 302 response
by elhar.mohamed＠gmail.com 31 Jul '24

31 Jul '24

Hello, I'm using the async_task_group_route for async module to handle some invites, when testing with sipp tools, I got some Dead call error message and after investigating It seems that for the same invite I sent multiple 302 message to sipp because the ACK sent by sipp was not handle before thr retr_timer1 & retr_timer2. Please can you help me to free the transaction after sending the first 302 ? Thank you in advance.

1 0

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

1 0

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

1 0

secfilter (and more modules) in kamailio kemi
by bazrafshan.m.h＠protonmail.com 28 Jul '24

28 Jul '24

Hi guys, some modules are not shown in Kamailio Kemi's documentation, such as secfilter. Is there any way to load them and use them in Python? a way is to simulate the secfilter but it is not a general solution. (Sorry in advance for the obvious question)

1 0

Job: System Engineer VoIP - Germany / HO
by Gerrit Jacobsen 25 Jul '24

25 Jul '24

VIER is looking for a System Engineer VoIP What VIER does VIER is a global contact center technology provider. VIER develops and runs its own technology in several data centers. You will be part of the Real-time Media Processing Team which is responsible for the telephony connections between carriers, applications and call-center agents. Your mission You are responsible for the support of the VoIP systems as well as the associated infrastructure Since you have sovereignty over the VoIP systems, you are also responsible for performing system upgrades and configurations. You have a lot of freedom to design, plan, and implement new VoIP system components for internal and external projects. You manage and implement change requests and are a specialist for special topics in the communication area. You show a high-sense of responsibility and foresight to keep systems running 24/7 You are prepared to work occasionally also in the evening hours to roll out new features Your skills You have several years experience in the day-to-day operation of telecom systems You have worked extensively with opensource VoIP software such as Kamailio / OpenSIPS, FreeSWITCH You can administer Linux systems such as RockyLinux and Debian You know SIP, RTP, RTCP, TLS, WebRTC protocols and can debug problems with Wireshark You have also worked with MySQL, Git You are communicative and like to work with team colleagues Further beneficial skills Experience with Ansible or Puppet for automation and configuration management. CheckMK, Homer, Hepic, Redis, CGRates, CDR Tools, Python Own VoIP software projects German language skills Please apply here: https://vier.jobs.personio.de/job/1638419?language=en&display=en

1 0

RTJSON: No Json for routing
by felipe.nunes＠comuniki.me 24 Jul '24

24 Jul '24

Good afternoon I have the following error in the application: [rtjson_routing.c:701]: rtjson_update_branch(): no json for routing This error occurs regardless of whether there is high demand or not and the application sending the data correctly. Can you help me with this? Felipe Nunes

2 1

ds_select from xhttp:request
by Christian Berger 24 Jul '24

24 Jul '24

Hi, we want to execute ds_select from an xhttp:request event route. The idea is to be able to get all active targets of a dispatcher set so our monitoring can check that via http which would be convenient for us. So are there any reasons why ds_select wouldn't work in an xhttp:request even route? Regards Christian Berger -- Christian Berger - berger(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

3 2

Listening on multiple TLS ports
by João Silva 23 Jul '24

23 Jul '24

Hello everyone! Is there a way to define a listen port range for TCP/TLS? Or any macro I can use to create the LISTEN commands? And finally is there a performance penalty for having multiple (2k) listening ports at the same time? I am in a strange situation where I need to register multiple credentials on a single SIP trunk, for each registration I need to use a different local port otherwise the trunk will simply overwrite the previous registrations. Furthermore, when I place a call on that trunk I have to be consistent with the port I used for registration for that specific credential. So I am looking into explicitly defining the port to be used per credential, UACREG allows me to use the contact_addr/socket for the registration part and for placing of outbound calls I can use $fsn to force the correct socket. Best regards, Joao

3 4

IMS PCSCF force source IP address
by Anthony Blandin 23 Jul '24

23 Jul '24

Hello, I am using Kamailio pcscf with ipsec. I have one ipsec listen address which is used by the UE to connect to the PCSCF and one other listen address on port 5060. During the registration until the ipsec is not established, the UE sends the requests to the ipsec listen address and the pcscf is using the other listen address to forward the sip request to the ICSCF. That's is ok for me. But when the ipsec is established after the 200 OK REGISTER, the UE sends a SUBSCRIBE but the pcscf forwards the SUBSCRIBE to the SCSCF using its ipsec listen address instead of the other listen address. I tried to use $fs to force the socket but without success. Do you know how to force the source IP address to be used on the pcscf? Thanks for your support. Regards, Anthony

1 0

mysql error after migration from 5.7.5 to 5.7.6
by Alejandro Guralnik 23 Jul '24

23 Jul '24

Hello, After upgraded from kamailio 5.7.5 to 5.7.6, when did start kamailio process, got the following error: WARNING: db_mysql [km_my_con.c:179]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring After downgrade to 5.7.5 dind't happened. (we don't use mysql ssl connection).

2 3

Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…

1 0

Best Practices for Scaling Kamailio in a Large VoIP Deployment
by Maria Jonas 23 Jul '24

23 Jul '24

Hello Kamailio Community, I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on: -Optimizing Kamailio’s configuration for high availability and performance. -Efficiently managing load balancing and failover. -Recommended hardware specifications for large-scale operations. -Common pitfalls to avoid when scaling up. Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced. Thank you! Maria <a href="https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…">aws</a>

1 0

SHM Memory leak?
by Ihor Olkhovskyi 23 Jul '24

23 Jul '24

Hello! I'm trying to get where all the SHM memory gone. For the moment I'm using these settings: -m 8192 -M 1024 which means 8G of SHM meory. Kamailio parameters are the following fork=yes children=8 tcp_children=12 enable_tls=yes enable_sctp=no tls_threads_mode=2 tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=10 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 And having ~1000 TLS/WSS clients I'm getting this after 3 days of running: { "jsonrpc": "2.0", "result": [ "shmem:fragments = 2796", "shmem:free_size = 3737947072", (3.7 G) "shmem:max_used_size = 4857418512", (4.8 G) "shmem:real_used_size = 4851987520", "shmem:total_size = 8589934592", "shmem:used_size = 4838988096" (4.8 G) ], "id": 984479 } Means half of SHM memory is gone When running kamcmd mod.stats all shm I'm getting (output truncated for the largest values) Module: core { ... tcpconn_new(1201): 130669280 (0.13 G) ... Total: 131657632 (0.13 G) } Module: sl { ... Total: 7520 } Module: siptrace { ... Total: 13520 } Module: rr { Total: 0 } Module: dialog { ... Total: 146080 } Module: permissions { ... Total: 62144 } Module: htable { .... Total: 3359552 (3.3 M) } Module: rtpengine { ... Total: 31456 } Module: textopsx { Total: 0 } Module: tsilo { ... Total: 75072 } Module: tm { .... Total: 3459504 (3.4 M) } Module: usrloc { .... Total: 1217616 } Module: pua_dialoginfo { ... Total: 8720 } Module: pua { ... Total: 150848 } Module: pike { ... Total: 9504 } Module: websocket { wsconn_add(198): 54139856 (54 M) ... Total: 54148096 (54 M) } Module: debugger { ... Total: 21872 } Module: tmx { Total: 512 } Module: kex { .... Total: 1888 } Module: tls { ... ser_malloc(364): 87246112 (87 M) ... Total: 87997168 (87 M) } Module: secfilter { ... Total: 768 } Module: exec { ... Total: 16 } Module: dispatcher { ... Total: 2992 } Module: cfgutils { ... Total: 48 } Module: app_python3 { ... Total: 32 } So, I'm guessing, where all the memory gone? Is there any way to get more detailed info on SHM? Kamailio 5.8.2 -- Thanks in advance, Ihor

2 6

Not responding to OPTIONS if existing tcp connection
by Jon Bonilla (Manwe) 22 Jul '24

22 Jul '24

Hi all version 5.7.6 Debian 12 I'm having issues with a kamailio server not sending 200 to some OPTIONS requests. It does send 200 to some subscribers but not 100% of the time to 100% of subs. I have enabled debug=3 in that server and this is what I found: If the log says that it found existing tcp connection and it's reusing it I can't see the 200ok in a sngrep capture. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/tcp_main.c:1722]: _tcpconn_find(): found connection by id: 2 <core> [core/tcp_main.c:2614]: tcpconn_send_put(): send from reader (360359 (18)), reusing fd <core> [core/tcp_main.c:2845]: tcpconn_do_send(): sending... <core> [core/tcp_main.c:2881]: tcpconn_do_send(): after real write: c= 0x7fb0faa893d8 n=384 fd=14 <core> [core/tcp_main.c:2882]: tcpconn_do_send(): buf=#012SIP/2.0 200 Keepalive#015#012Via: SIP/2.0/TCP 10.35.190.105:5060;branch=z9hG4bK15ebf764;rport=55894;received=1.1.1.186#015#012From: "Unknown" <sip:contesta01@10.35.190.105>;tag=as5311c3a7#015#012To: <sip: 1.1.1.138>;tag=054524a74bcccf444a0f08f67c5b7657.d07b5401#015#012Call-ID: 15dcd38c46f915576ded132a11d8ff4c@10.35.190.105:5060#015#012CSeq: 102 OPTIONS#015#012Server: aaa SBC#015#012Content-Length: 0#015#012#015#012 <core> [core/receive.c:531]: receive_msg(): request-route executed in: 302 usec On the other side, if kamailio doesn't find an existing tcp connection (at least nothing in the logs about that) I can see the 200 ok in sngrep. Example: <core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply <core> [core/receive.c:531]: receive_msg(): request-route executed in: 295 usec All subs use tcp. And the kamailio.cfg is very simple regarding OPTIONS: Sanity, check maxfwd and sl_send_reply(200). Any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

Drouting using TLS and domain names
by SAMUEL MOYA TINOCO 19 Jul '24

19 Jul '24

Hi everyone, I'm trying to configure drouting module but i'm not sure how to configure dr_gateways table to use domain names and TLS instead of IPs How should I do it? In dispatcher module is by using sips:domain but when I configure it like this I have problems to send invites (maybe I'm doing something wrong). Does anyone know how to configure it? Thanks in advance for your help Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> [cid:image001.png@01DAD156.D8CFF170] Soluciones inteligentes para la autonomía personal

2 3

Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 18 Jul '24

18 Jul '24

Hello all, I am using a very simple scritp with corex module enabled . Normal calls works fine, but a scenario where INVITE does not receive response and the call is released with SIP 408 (request timeout) leads to memmory leak. I verified it making a lot of calls with this scenario and printing the shared memory (using kamcmd core.shmmem) . Its ocurred with modparam("corex", "nio_intercept", 1). If I modify to modparam("corex", "nio_intercept", 0) the memmory leak does not ocurr anymore. The problem also occurs in another scenario where the SIP Cancel message is not responded to. My script has this event route # event route event_route[network:msg] { xlog("L_INFO", "Dentro do Event_Route \n"); if (is_incoming()) { xlog("L_INFO", "Received message '$mb' \n"); $avp(msg) = $mb; xinfo("avp(msg) = $avp(msg)"); } else { xlog("L_INFO", "Sending message '$mb' \n"); $avp(msg) = $mb; }; } Test done wtih 4000 calls: [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 900", "shmem:free_size = 6180368", "shmem:max_used_size = 66980272", "shmem:real_used_size = 60928496", "shmem:total_size = 67108864", "shmem:used_size = 54884944" ], "id": 393115 } [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { create_avp(178): 52257504 counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 52687808 } I attached my kamailio script used and a log with one call using the command "kamcmd corex.shm_summary" with modparam("corex", "nio_intercept", 1) and modparam("corex", "nio_intercept", 0) Regards, Vanderlei

3 6

Re: Difference betwwen TM functions in KEMI
by Duarte Rocha 18 Jul '24

18 Jul '24

Hello Alex, You're totally right, that was the issue. Thank you so much for the help. Helpful as always. Cheers, Duarte Rocha

1 0

Difference betwwen TM functions in KEMI
by Duarte Rocha 17 Jul '24

17 Jul '24

Hello all, I'm using Kamailio with native script and KEMI in Javascript. I noticed that functions from the TM module can have different results if I call them on KEMI or Native script. Is this intended behaviour? Don't they share the state? In order to test this i made the following code : KEMI Code : function ksr_request_route() { try{ if ( KSR.is_method("BYE") ) { print.error("BYE Request KEMI BYE") KSR.route("exists_test") if(KSR.tm.t_exists()) { print.error("KEMI : Transaction exists") } else { print.error("KEMI : Transaction does not exist") } Native Code : route[exists_test] { if( t_exists()) { xerr("Native Script : Transaction exists \n"); } else { xerr("Native Script : Transaction does not exist \n"); } } I've also tried to change function call order but every time I get "Transaction does not exist" on native and "Transaction Exists" on KEMI. Is this intended behaviour? Should I only call tm functions in native or Kemi in order to have consistency ? Thanks in advance, Best Regards,

2 1

Dns record not alone?
by Pyry Aaltonen 17 Jul '24

17 Jul '24

Hi, I recently updated my testing env to 5.8.1 and after that I’ve been receiving dns_cache.c warning “record not alone”. I found out that it’s been added to quite recent commit: https://github.com/kamailio/kamailio/commit/d8a35b3b6c837b36779e232b65fce61… And that’s probably the why I haven’t seen it before. As far as I know, I have no other problem with dns or anything else than the warning appearing in the logs 😊 I can’t quite get what the warning is about? Clearly refers to dns queries, and with debug I found out that maybe its somehow related to ip which has several domains to pointing it (and queried by one of those domains by this kamailio). Can it be something with the PTR-record? And this warning comes ones every hour, so probably when some ttl expires, and meanwhile the cache is ok with the situation. And as a background, I haven’t done any specific dns-configurations in my config, so settings should be pretty much in defaults. Any advice what I should fix here to get the warning disappear? -Pyry

2 1

Add a Header to CANCEL request
by sadik.oualla.mohamed＠gmail.com 17 Jul '24

17 Jul '24

Hello all, I am facing an issue with Kamailio when trying to add custom headers to CANCEL requests. My current configuration adds the "Reason" header perfectly: ``` if(!is_present_hf("Reason")){ $var(cause_request) = 31; append_hf("Reason: Q.850;cause=$var(cause_request)\r\n", "CSeq"); msg_apply_changes(); } ``` However, when I attempt to add headers like "X-Reason" or "TestHeader", they are not added: ``` if(!is_present_hf("Reason")){ $var(cause_request) = 31; append_hf("X-Reason: Q.850;cause=$var(cause_request)\r\n", "CSeq"); # or append_hf("TestHeader: Q.850;cause=$var(cause_request)\r\n", "CSeq"); msg_apply_changes(); #nothing changes in the relayed CANCEL request :-/ } ``` I've read in this discussion "https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…" that this might work if sending CANCEL in stateless mode, but I want to keep it stateful. Any suggestions on how to achieve this? Regards, Mohamed.

1 0

Changing ksr_tcp_accept_iplimit from default 1024
by Who AmI 16 Jul '24

16 Jul '24

Hi list, I hope you are all well? I am currently loadtesting some services internally and I am getting an odd error around TCP connections from the same source IP below: "/usr/sbin/kamailio[1806789]: CRITICAL: <core> [core/tcp_main.c:4459]: handle_new_connect(): hit the limit of connections per source IP (my_ip:39043) - rejecting" Its from the core the error so I was looking at the cookbooks and documentation but cannot get around this specific limit or find a configuration to tweak that fixes it. The limit only affects TCP (will affect TLS as well but I haven't got that far) and is 1024 it seems. Kamailio version is: 5.7.6 OS: Ubuntu 22.04 Things I have done so far: I have set limitNOFILE=65535 in the service file Set the tcp/tls_max_connections at 8192 (default is 2048) Increase the tcp_rd_buf_size to 16k+ The service I am using to loadtest is SIPp with flags "-t tn" for TCP multisocket as this service will have much more than 1024 concurrent TCP connections to it from a single source IP. I think I have missed something basic here as the devel and the source code references: "ksr_tcp_accept_iplimit" here at line 541: https://github.com/kamailio/kamailio/blob/master/src/main.c#L541 I cannot find a way to change this with a configuration line it seems, am I being blind or is my only option to build from source and change this? Many Thanks, John.

2 2

Weird telephone-event payload issue - need help
by Tim Bowyer 16 Jul '24

16 Jul '24

Hello Everyone, I've been grappling with this issue for some months, time to raise the white flag. Essentially I need to work out a way to configure Kamailio/RTPengine to normalise telephone-events to 101 earlier in the piece. 1. Call comes in from Carrier with INVITE w/SDP, telephone-event payload of 96. 2. INVITE sent to Asterisk with the same SDP, payload = 96 3. INVITE sent from Asterisk to Client with payload = 101 * 200 OK w/SDP from Client has payload = 101 * 200 OK w/SDP to Kamailio has payload = 96 * 200 OK w/SDP to Carrier has payload = 96 4. Asterisk (with Direct Media enabled) sends INVITE to Kamailio with payload = 96, and new media IP of Client 5. Kamailio passes this onto Carrier with media IP of RTPengine 6. 200 OK w/SDP from Carrier has payload = 96 sent to Kamailio 7. Kamailio passes this 200 OK to Asterisk, which responds with ACK and doesn't do anything further +---------+ +----------+ +---------+ +---------+ | Carrier | <----1-----> | Kamailio | <-----2----> | Asterisk| <----3------>| Client | | | | | | | | | +---------+ +----------+ +---------+ +---------+ ^ | | | | | |3| | | | | | +------5---------> +-----------+ <-----------------+------------4-----------+ | RTPengine | +-----------+ Big issue is, the Client is expecting telephone-events with a payload of 101, whereas RTPengine is passing it 96 from the Carrier. * Inbound DTMF only works at client-side if Asterisk stays in path (far from ideal). * The issue is exacerbated when calls are sent back out to the same or another carrier (forward scenario). * There's nothing I can seem to do in Asterisk to change how it handles DTMF in a direct media/re-invite setting. Any ideas? Cheers, Tim Bowyer

3 12

how to upgrade kamailio minor release from 5.7.5 to 5.7.6
by sadik.oualla.mohamed＠gmail.com 15 Jul '24

15 Jul '24

I am currently using Kamailio version 5.7.5, which I installed from the source code. I would like to upgrade to version 5.7.6 but have been unable to find documentation specifically addressing minor upgrades between these minor releases. I have found only wikis detailing the process for upgrading for major releases (like 5.7.x to 5.8.0). Could you please point me to the documentation for my current need or share with me steps on how to perform this minor upgrade? Thank you. Cheers.

3 4

How to get the rtpengine IP used by the current call among multiple rtpengines
by mm e 15 Jul '24

15 Jul '24

modparam("rtpengine", "rtpengine_sock","udp:192.168.1.3:12221=2 udp:192.168.1.1:12222=1 udp:192.168.1.5:12221=7") When rtpengine_sock is set to multiple rtproxies mode, how to get the id or ip of the rtpengine used in the current call? There is no information returned which rtpengine

1 0

How to get the id or ip in the multiple rtproxies for LB with weights。
by mm e 15 Jul '24

15 Jul '24

modparam("rtpengine", "rtpengine_sock","udp:192.168.1.3:12221=2 udp:192.168.1.1:12222=1 udp:192.168.1.5:12221=7") When rtpengine_sock is set to multiple rtproxies mode, how to get the id or ip of the rtpengine used in the current call? There is no information returned which rtpengine kamailio selected for the call.

1 0

Re: (no subject)
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi Brett! From what I could understand, my client (a Carrier) assigns services and products to its customers identified by trunkgroups (depending on the customer and the service requested, as well as the ingress trunkgroup the call will be forwarded to a specific egress trunkgroup). . The INVITE is sent by customer to the SBC (ingress), where some kind of process (which i have no details) will decide what IP:Port to forward the messages to Kamailio with additional infos (Kamailio is listening on different NICs/IPs/Ports). Then, Kamailio will check on the (external) routing engine the route to take and forwards to the outbound endpoint. Depending on the IP:Port that Kamailio receives the call (SBC may choose one NIC or the other) the request is then received by Kamailio and verified/processed by the routing engine replying back to kamailio with the route to take, and then Kamailio forwards to the outbound endpoint (egress SBC...). It is a bit confusing, and I want to simplify things as much as possible, but apparently the old/current solution is set this way and they pretend to keep the new solution as similar as possible to the old/current solution. This is mostly due to the billing process, which is very (overly?) complex and they do not pretend to change it. IMHO, the routing engine should be much simpler but it is a much more complex process the Carrier currently has...and wishes to keep. (sorry for posting without a subject. I did, however, re-posted with corrected subject) Atenciosamente / Kind Regards / Cordialement, *Sérgio Charrua* On Fri, Jul 12, 2024 at 8:11 PM Brett Nemeroff <brett(a)voicefoxtelephony.com> wrote: > Is there a reason you have to identify the trunk group based on received > IP/Port? I've seen this as a requirement from some more antiquated systems > and it's always a pain. It's always better to use source IP or even a trunk > prefix (dialed number prefix) which isn't really secure, but works when > security isn't an issue. > > If you have to do it that way, I'd probably turn the $Ri/$Rp into some > sort of cache key like $Ri_$Rp and map it to something like a dispatcher > setid. > > There may be a more modern way to do this, but I don't think there is. > > Good luck! > -Brett > > > On Fri, Jul 12, 2024 at 1:40 PM Sergio Charrua via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> Hi all! >> >> for a project under development, one requirement is to be able to >> forward/relay the call to a specific destination depending on which IP >> Address and Port the call was received by Kamailio. >> This also means that Kamailio will be listening on multiple IP Address >> and Ports >> listen=udp:IP_1:port_1 # trunkgroup 1 >> listen=udp:IP_2:port_2 # trunkgroup 2 >> listen=udp:IP_3:port_3 # trunkgroup 3 >> >> I know the pv $Ri and $Rp can be used to identify from which IP address >> and port the message reached Kamailio and having those details, depending >> on the value, I can define some conditions that allows Kamailio to relay >> the call to different destination endpoints. >> For example: >> - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC >> address A.B.C.D >> - but calls from +32 to +39 received on trunkgroup2 need to go to Germany >> on SBC address 1.2.3.4 (as it is cheaper) >> >> Is there a better way of doing this without using IF/THEN/ELSE or SWITCH >> blocks and "hard code" destination SBC addresses? >> I have read the DRouting module's documentation but not sure if it could >> help and how... >> >> Any suggestions? >> >> >> >> Atenciosamente / Kind Regards / Cordialement, >> >> >> *Sérgio * >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to >> the sender! >> Edit mailing list options or unsubscribe: >> > > > -- > > ======================================================== > Brett Nemeroff > Voice Fox Telephony LLC > Office: 512-670-8369 > Email: brett(a)voicefoxtelephony.com >

1 0

Change the To Header in the failure_route
by sadik.oualla.mohamed＠gmail.com 12 Jul '24

12 Jul '24

Hello, I need help with modifying the "To" header in the "failure_route[ID]{...}". Specifically, I want to add the "to tag" parameter from the previous SIP messages of the current dialog(e.g., 180 Ringing, PRACK or final responses). Here's the scenario: 1. Kamailio receives and relays the SIP messages: INVITE, 180 Ringing, PRACK, 200 OK, 486 Busy Here. 2. In my case, the 180 Ringing includes the "to tag" parameter. 3. When Kamailio receives the 486 response, it triggers "failure_route[ID]" using "t_on_failure(ID)" method. 4. I extract the SIP response code and reason from the final response of the INVITE transcation in this failure_route. 5. After processing and applying some instructions, I send a reply using "t_send_reply" with the code and reason extracted. The problem: - The failure_route[ID] holds the initial INVITE, which has no "to tag" parameter. - When I send a reply using "t_send_reply()", Kamailio generates a random "to tag" different from the previous SIP messages, which confuses my UAC. I think adding the "to tag" parameter manually might be a solution, but I'm not sure if I'm doing it correctly or if there is a better approach. I also tried using uac_replace_to() method, but it doesn't work in the failure_route[ID]. How can I make sure the reply in failure_route[ID] uses the correct "to tag" from the previous SIP messages? Are there any other possibilities to achieve this? Any assistance would be appreciated. Thank you!

5 11

Kemi Python3 throws a TypeError
by Michel Pelletier 12 Jul '24

12 Jul '24

Hello, Please help. I am using Kemi with the python3 module. I get random TypeError exceptions thrown within the Kamailio code. For example I have the following code: event = str(KSR.pv.get("$hdr(Event)")) log.logInfo( "KamailioStateful.ksr_request_route_loopback(): " + "Passing the request to the presence " + "publish handler for event: " + event ) rc = KSR.presence.handle_publish() if rc != 1: log.logError( "KamailioStateful.ksr_request_route_loopback(): " + "The presence publish handler returned " + "an error code: " + str(rc) ) This is what I see in the logs: 21(120) ERROR: PY3 {PUBLISH}: app_python3 [python_support.c:167]: python_handle_exception(): apy_exec: ksr_request_route((null)): Unhandled exception in the Python code: TypeError: expected bytes, str found The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 91, in ksr_request_route return ksf.ksr_request_route(msg, "Loopback") ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 100, in ksr_request_route return self.ksr_request_route_loopback(msg) ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ File "/etc/kamailio/kamailio_stateful.py", line 240, in ksr_request_route_loopback rc = KSR.presence.handle_publish() ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ SystemError: <built-in function handle_publish> returned a result with an exception set If I put the offending line within a Try/Except it hides it but I would rather not see it not happen at all. Any Kemi Python3 users experiencing something similar? What could be causing this? Regards, Michel Pelletier

2 1

routing call depending on which IP address SIP message reached Kamailio
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi all! For a project under development, one requirement is to be able to forward/relay the call to a specific destination depending on which IP Address and Port the call was received by Kamailio. This also means that Kamailio will be listening on multiple IP Address and Ports listen=udp:IP_1:port_1 # trunkgroup 1 listen=udp:IP_2:port_2 # trunkgroup 2 listen=udp:IP_3:port_3 # trunkgroup 3 I know the pv $Ri and $Rp can be used to identify from which IP address and port the message reached Kamailio and having those details, depending on the value, I can define some conditions that allows Kamailio to relay the call to different destination endpoints. For example: - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC address A.B.C.D - but calls from +32 to +39 received on trunkgroup2 need to go to Germany on SBC address 1.2.3.4 (as it is cheaper) Is there a better way of doing this without using IF/THEN/ELSE or SWITCH blocks and "hard code" destination SBC addresses? I have read the DRouting module's documentation but not sure if it could help and how... Any suggestions? Atenciosamente / Kind Regards / Cordialement, *Sérgio *

2 3

(no subject)
by Sergio Charrua 12 Jul '24

12 Jul '24

Hi all! for a project under development, one requirement is to be able to forward/relay the call to a specific destination depending on which IP Address and Port the call was received by Kamailio. This also means that Kamailio will be listening on multiple IP Address and Ports listen=udp:IP_1:port_1 # trunkgroup 1 listen=udp:IP_2:port_2 # trunkgroup 2 listen=udp:IP_3:port_3 # trunkgroup 3 I know the pv $Ri and $Rp can be used to identify from which IP address and port the message reached Kamailio and having those details, depending on the value, I can define some conditions that allows Kamailio to relay the call to different destination endpoints. For example: - calls from +32 to + 39 received on trunkgroup1 needs to go to Italy SBC address A.B.C.D - but calls from +32 to +39 received on trunkgroup2 need to go to Germany on SBC address 1.2.3.4 (as it is cheaper) Is there a better way of doing this without using IF/THEN/ELSE or SWITCH blocks and "hard code" destination SBC addresses? I have read the DRouting module's documentation but not sure if it could help and how... Any suggestions? Atenciosamente / Kind Regards / Cordialement, *Sérgio *

1 0

deb package build error (v5.7.6) [error: debian/rules binary subprocess returned exit status 2]
by Denys Pozniak 12 Jul '24

12 Jul '24

Hello! I'm trying to build standard deb packages for Kamailio v5.7.6 for Debian 12.6, but I get an error. By the way, there are no problems with building v5.7.2 ... #git clone --single-branch --branch 5.7.6 https://github.com/kamailio/kamailio.git /usr/src/kamailio #cd /usr/src/kamailio && ln -s pkg/kamailio/deb/bookworm debian #make cfg && RADCLI=1 make deb .... Makefile.defs defs skipped make[6]: 'libkamailio_ims.so.0.1' is up to date. gcc -shared -g -m64 -Wl,-O2 -Wl,-E -Wl,-z,relro -pthread -rdynamic "-ldl" -Wl,-Bsymbolic-functions cmd.o ims_ipsec_pcscf_mod.o ipsec.o sec_agree.o spi_gen.o spi_list.o spi_list_tests.o -lm -lmnl -L../../lib/ims/ -lkamailio_ims -Wl,-rpath,/usr/src/kamailio/src/lib/ims -o ims_ipsec_pcscf.so make[4]: Leaving directory '/usr/src/kamailio/src' make[3]: Leaving directory '/usr/src/kamailio' make[2]: Leaving directory '/usr/src/kamailio' *dpkg-buildpackage: error: debian/rules binary subprocess returned exit status 2* make[1]: Leaving directory '/usr/src/kamailio/src' -- BR, Denys Pozniak

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New Member Introduction
by Maria Jonas 11 Jul '24

11 Jul '24

Hi everyone, I am new to this forum and excited to be here. I'm eager to learn more about Kamailio and its features and I'm looking forward to participating in discussions here. Could someone please guide me on how to ask questions here ? Where should I post if I have a query ? Looking forward to your advice and connecting with you all. Thanks,

2 1

usrloc ka_interval timer is not correct in kamailio 5.8
by Michal Popovič 11 Jul '24

11 Jul '24

Hello everybody, I am using kamailio in AWS behind load balancer and after upgraded from version 5.7 to 5.8 kamailio is not able to keep connection open. After quick investigation I found out that usrloc module got new parameter ka_interval that replaced timer_interval. Change is here. https://github.com/kamailio/kamailio/commit/53ace443020075f7ecd12c72f154193… Problem is that this interval is not reliable, as it is most probably still called only when timer_interval is executed. If I set values to timer_interval = 30 ka_interval = 30 ka_randomize = 20 Maximum time for keep-alive is not 30+20 seconds but 30+30+20 seconds. That is too late for AWS load balancer. I have fixed it by enabling tcp keepalives on kernel level. Is there any way how to specify exact time in version 5.8? Thanks. Michal

1 0

HTTP Client configuration path
by Sergio Charrua 11 Jul '24

11 Jul '24

Hi all! I am using HTTP_CLIENT module, and have created a http configuration file. The module's config_file parameter is set as: modparam("http_client", "config_file", "/etc/kamailio/http_client.cfg") and it works OK on my DEV environment, because there is a symlink /etc/kamailio pointing to physical folder /usr/local/etc/kamailio/ (using Kamailio 5.7.4) However in our integration environment (also using Kamailio 5.7.4), which is identical to production, and due to limitations on access rights, we cannot create a symlink /etc/kamailio pointing /usr/local/etc/kamailio/, so all cfg files are in the installation folder. So I thought that the config_file parameter should be a relative path, starting with " ./ " or by just setting the config file name, without path (as per documentation at http_client (kamailio.org) <https://kamailio.org/docs/modules/5.1.x/modules/http_client.html#idp38265140> . The issue is that even though the file http_client.cfg is present in /usr/local/etc/kamailio/, as well as other cfg files, the cfg_parser fails to load it: Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: <core> > [core/cfg_parser.c:632]: cfg_parser_init(): cfg_parser: Unable to open file > './http_client.cfg' > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: http_client > [curlcon.c:782]: http_client_load_config(): Failed to init http_client > config file parser > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: http_client > [http_client.c:307]: mod_init(): Failed to load http_client connections > from [./http_client.cfg] > Jul 11 09:32:09 lab-kamailio kamailio[44114]: ERROR: <core> > [core/sr_module.c:1030]: init_mod(): Error while initializing module > http_client (/usr/local/lib64/kam > ailio/modules/http_client.so) I have set the parameter to > modparam("http_client", "config_file", "./http_client.cfg") and also to > modparam("http_client", "config_file", "http_client.cfg") but it only works if I add it as : > modparam("http_client", "config_file", > "/usr/local/etc/kamailio/http_client.cfg") or by adding > chroot="/usr/local/etc/kamailio" in the main cfg script, which was my final solution. Also checked for permissions, and all is fine. Note that this only seems to happen with http_client config_file parameter. I have other include commands in the main cfg file with relative paths and all works fine. I have also analysed the cfg_parser.so file, which loads the configuration file for http_client module, but could not find anything wrong with it (at 1st glance....) What am I doing wrong? Any clue? Atenciosamente / Kind Regards / Atentamente / Cordialement, *Sérgio *

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SIP Question: B2BUA not ACKing 200 OK response
by Ghulam Mustafa 11 Jul '24

11 Jul '24

Hi Community, We are setting up a new provider in our VoIP setup for outgoing calls (FreeSWITCH --> Kamailio --> Provider). When the provider sends a 200 OK in response to our INVITE, FreeSWITCH's SIP stack discards the message, and we are unable to determine the cause of this behavior. I am sharing the SIP message exchange below. Could any SIP experts assist us in understanding why FreeSWITCH is not sending an ACK in response to the 200 OK message? We understand that the issue is not related to Kamailio but seems to be within the SIP stack or FreeSWITCH itself. Below are the actual INVITE and its response, which was discarded by FreeSWITCH: send 1203 bytes to tcp/[10.xxx.x.30]:5060 at 15:48:23.001054: ------------------------------------------------------------------------ INVITE sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.xxx.x.10:5080;rport;branch=z9hG4bKD8p0KmvB8ZKma Max-Forwards: 69 From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx> Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Contact: <sip:gw+commsgroup@10.xxx.x.10:5080;transport=tcp;gw=commsgroup> User-Agent: FreeSWITCH-mod_sofia/1.10.11-release~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: 347 X-FS-Support: update_display,send_info Remote-Party-ID: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10 >;party=calling;screen=yes;privacy=off v=0 o=Z 5395246197255529491 693195311 IN IP4 10.xxx.x.225 s=Z c=IN IP4 10.xxx.x.225 t=0 0 m=audio 55319 RTP/AVP 106 9 98 101 0 8 3 a=rtpmap:106 opus/48000/2 a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 a=rtpmap:98 telephone-event/48000 a=fmtp:98 0-16 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-mux tport.c:2316 tport_set_secondary_timer() tport(0x150e6c0291c0): reset timer nta.c:8400 outgoing_send() nta: sent INVITE (85714459) to tcp/10.xxx.x.30:5060 tport.c:4191 tport_pend() tport_pend(0x150e6c0291c0): pending 0x150e6c0c52f0 for tcp/10.xxx.x.30:5060 (already 2) nua_session.c:4131 signal_call_state_change() nua(0x150d9c107460): call state changed: init -> calling, sent offer soa.c:1268 soa_get_local_sdp() soa_get_local_sdp(static::0x150e6c0c0210, [0x150e91772798], [0x150e91772790], [(nil)]) called nua_stack.c:299 nua_stack_event() nua(0x150d9c107460): event i_state INVITE sent nua_stack.c:389 nua_application_event() nua: nua_application_event: entering nua.c:1170 nua_handle_unref_user() nua nua_handle_unref_user(0x150d9c107460): entering nua.c:365 nua_handle_magic() nua: nua_handle_magic: entering nua_stack.c:558 nua_signal() nua(0x150d9c107460): sent signal r_handle_unref nua.c:1157 nua_unref_user() nua: nua_unref_user: entering 2024-07-09 15:48:22.985772 100.00% [DEBUG] sofia.c:7493 Channel sofia/external/+86xxxxxxxxxxx entering state [calling][0] nua_stack.c:599 nua_stack_signal() nua(0x150d9c107460): recv signal r_handle_unref nua.c:365 nua_handle_magic() nua: nua_handle_magic: entering nua_stack.c:558 nua_signal() nua((nil)): sent signal r_unref nua.c:1170 nua_handle_unref_user() nua nua_handle_unref_user(0x150d9c107460): entering nua_stack.c:599 nua_stack_signal() nua((nil)): recv signal r_unref nua_stack.c:558 nua_signal() nua(0x150d9c107460): sent signal r_handle_unref nua_stack.c:599 nua_stack_signal() nua(0x150d9c107460): recv signal r_handle_unref nua.c:1157 nua_unref_user() nua: nua_unref_user: entering nua_stack.c:558 nua_signal() nua((nil)): sent signal r_unref nua_stack.c:599 nua_stack_signal() nua((nil)): recv signal r_unref 2024-07-09 15:48:22.985772 100.00% [DEBUG] switch_core_state_machine.c:659 (sofia/external/+86xxxxxxxxxxx) State CONSUME_MEDIA 2024-07-09 15:48:22.985772 100.00% [DEBUG] switch_core_state_machine.c:659 (sofia/external/+86xxxxxxxxxxx) State CONSUME_MEDIA going to sleep tport.c:2796 tport_wakeup() tport_wakeup(0x150e6c0291c0): events IN tport.c:2900 tport_recv_event() tport_recv_event(0x150e6c0291c0) tport.c:3237 tport_recv_iovec() tport_recv_iovec(0x150e6c0291c0) msg 0x150e6c0cb390 from (tcp/10.xxx.x.30:5060) has 352 bytes, veclen = 1 recv 352 bytes from tcp/[10.xxx.x.30]:5060 at 15:48:23.004029: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/TCP 10.xxx.x.10:5080;rport=37555;branch=z9hG4bKD8p0KmvB8ZKma;received=10.xxx.x.10 From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx> Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Server: Kamailio 5.8 Content-Length: 0 tport.c:3055 tport_deliver() tport_deliver(0x150e6c0291c0): msg 0x150e6c0cb390 (352 bytes) from tcp/10.xxx.x.30:5060/sip next=(nil) nta.c:3378 agent_recv_response() nta: received 100 Trying for INVITE (85714459) nta.c:3445 agent_recv_response() nta: 100 Trying is going to a transaction nta.c:9697 outgoing_estimate_delay() nta_outgoing: RTT is 3.038 ms tport.c:4253 tport_release() tport_release(0x150e6c0291c0): 0x150e6c0c52f0 by 0x150e6c0be9a0 with 0x150e6c0cb390 (preliminary) tport.c:2316 tport_set_secondary_timer() tport(0x150e6c0291c0): reset timer nta.c:5904 incoming_reclaim_queued() incoming_reclaim_all((nil), (nil), 0x150e91973a50) nta.c:7270 _nta_incoming_timer() nta_incoming_timer: 0/0 resent, 0/0 tout, 0/0 term, 2/3 free nta.c:1310 agent_timer() nta: timer not set tport.c:2796 tport_wakeup() tport_wakeup(0x150e6c0291c0): events IN tport.c:2900 tport_recv_event() tport_recv_event(0x150e6c0291c0) tport.c:3237 tport_recv_iovec() tport_recv_iovec(0x150e6c0291c0) msg 0x150e6c0c9d90 from (tcp/10.xxx.x.30:5060) has 1195 bytes, veclen = 1 recv 1195 bytes from tcp/[10.xxx.x.30]:5060 at 15:48:27.070278: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TCP 10.xxx.x.10:5080;received=10.xxx.x.10;rport=37555;branch=z9hG4bKD8p0Km From: "+1xxxxxxxxxx" <sip:+1xxxxxxxxxx@10.xxx.x.10>;tag=ZSvBBtypKypDS To: <sip:+86xxxxxxxxxxx@35.xxx.xxx.xxx >;tag=7bQ0OSl9Dl-39455-57712329-1057554-14 Call-ID: 21b7b9b5-b8a5-123d-63a6-ecf4bbd64f6c CSeq: 85714459 INVITE Contact: <sip:127.0.0.8;line=sr-H46V9.fbsXNDlkfEJjGbujK3J4aDo1GzP5GOJ8hcsXwV9LAWw1abHkqWlXczwL7*> Record-Route: <sip:13x.xxx.xxx.xx;transport=tcp;lr;ftag=ZSvBBtypKypDS;did=cbd.e581;nat=yes> Supported: sdp-anat Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY Require: timer Session-Expires: 2000;refresher=uas Server: Mediant VE SBC/v.7.40A.250.265 P-Asserted-Identity: <sip:4006100666@hk.chinatelecom.com;user=phone> Accept: application/sdp, application/isup, multipart/mixed, application/dtmf, application/dtmf-relay Content-Type: application/sdp Content-Length: 219 User-Agent: Kamailio 5.8 v=0 o=- 846721169 1919912032 IN IP4 13x.xxx.xxx.xx s=- c=IN IP4 13x.xxx.xxx.xx t=0 0 m=audio 49708 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:49709 tport.c:3055 tport_deliver() tport_deliver(0x150e6c0291c0): msg 0x150e6c0c9d90 (1195 bytes) from tcp/10.xxx.x.30:5060/sip next=(nil) nta.c:3378 agent_recv_response() nta: received 200 OK for INVITE (85714459) nta.c:3494 agent_recv_response() nta: 200 OK was discarded Any insights or suggestions would be greatly appreciated, Specifically, why is the 200 OK response being discarded?. Thank you! Regards, Mustafa

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Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 11 Jul '24

11 Jul '24

Hi, Daniel. Thank you for the quick response! I will update you after testing the fix. Regards, Vanderlei

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Re: Possible Kamailio memmory leak - version 5.8.2
by Vanderlei Torres Batistela 10 Jul '24

10 Jul '24

Hello Henning, Thank you for your quick response. The increase happens in the core module (create-avp). See bellow: [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { create_avp(178): 52257504 <---------------------------- counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 52687808 <------------------ } [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 900", "shmem:free_size = 6180368", "shmem:max_used_size = 66980272", "shmem:real_used_size = 60928496", "shmem:total_size = 67108864", "shmem:used_size = 54884944" ], "id": 393115 } These reports above are shown after the end of calls (4000) where the outgoing INVITE is not answered. In normal calls the issue does not occur and the reports are shown below: [root@labcom070 kamailio]# kamcmd mod.stats all shm Module: core { counters_prefork_init(211): 53760 cfg_clone_str(132): 112 cfg_shmize(221): 832 main_loop(1381): 16 init_pt(104): 16 init_pt(103): 16 init_pt(102): 6224 cfg_register_ctx(47): 96 init_tcp(5197): 8192 init_tcp(5191): 32768 init_tcp(5182): 16 init_tcp(5175): 16 init_tcp(5167): 16 init_tcp(5161): 16 init_tcp(5149): 16 init_avps(92): 16 init_avps(91): 16 init_dst_blocklist(435): 16384 init_dst_blocklist(427): 16 timer_alloc(494): 96 init_dns_cache(368): 16 init_dns_cache(359): 16384 init_dns_cache(351): 16 init_dns_cache(343): 16 init_timer(264): 16 init_timer(263): 16384 init_timer(262): 16 init_timer(261): 16 init_timer(250): 16 init_timer(218): 16 init_timer(207): 278544 init_timer(206): 16 init_timer(194): 16 cfg_child_cb_new(832): 64 sr_cfg_init(371): 16 sr_cfg_init(364): 16 sr_cfg_init(356): 16 sr_cfg_init(344): 16 sr_cfg_init(332): 16 ksr_shutdown_phase_init(123): 16 rpc_hash_add(118): 16 qm_shm_lock_init(1463): 16 Total: 430304 } [root@labcom070 kamailio]# kamctl stats shmem { "jsonrpc": "2.0", "result": [ "shmem:fragments = 6", "shmem:free_size = 64143184", "shmem:max_used_size = 4070816", "shmem:real_used_size = 2965680", "shmem:total_size = 67108864", "shmem:used_size = 2627856" ], "id": 683948 } Regars, Vanderlei

2 1

Intercepting OPTIONS from nathelper
by Denys Pozniak 10 Jul '24

10 Jul '24

Hello! I need advice on how to intercept local SIP OPTIONS requests from the nathelper module for some adjustments? As far as I tested, it does not fall into the *event_route[tm:local-request]* Thanks in advance! # kamailio -v version: kamailio 5.7.1 (x86_64/linux) 1cf389-dirty -- BR, Denys Pozniak

2 5

handle_subscribe(), handle_publish() and DB usage
by Володимир Іванець 04 Jul '24

04 Jul '24

Hello everyone! According to the documentation functions handle_subscribe() and handle_publish() update information not only in memory but also in the database. Isn't there a way to disable database usage for this functionality? Setting subs_db_mode to 0 does not make a change. Thank you!

1 0

Contact does not exist in PCSCF usrloc error in log
by Shantanu K Roy 03 Jul '24

03 Jul '24

Hello, We are trying to integrate Kamailio as the IMS server with our 5G core emulator by following the steps listed out at https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. In our case, we have our own 5G core emulator instead of the Open5GS core. We are using sipp scripts to generate the SIP calls. The PCRF component of the 5G core is emulated by us and is capable of handling Diamater calls from Kamailio over the Rx interface. We have got the setup working to the point where the SIP Register calls are landing on the PCSCF which in turn is sending out AA Requests to PCRF and the PCRF is responding. However there is an internal error in the Kamailio log "4(3682133) ERROR: ims_qos [ims_qos_mod.c:1305]: w_rx_aar_register(): This contact does not exist in PCSCF usrloc - error in cfg file" Please help us in identifying if we are missing anything in the Kamailio set up. The packet capture and syslog excerpts from the VM where Kamailio is running are attached; here we are trying to perform SIP Registration for three SIP endpoints 001010123456791, 001010123456792 and 001010123456793. If there is any additional information required, please let us know. Regards, Shantanu Kumar Roy

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Kamailio v5.6.6 Released
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, Kamailio SIP Server v5.6.6 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.6. Note that 5.6 is now the third last stable branch, v5.6.6 being the last planned released in 5.5.x series. The latest two stable branch are 5.7 and 5.8, with v5.8.2 being released a while ago. For more details about version 5.6.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-6-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training -- asipto.com

1 0

Releasing v5.6.6
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, I am considering to release Kamailio v5.6.6 (out of branch 5.6) later this week (likely on Wednesday, July 3, 2024). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. This is going to be the last release out of branch 5.6 to mark the end of official/packaging maintenance. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

Extracting and Handling Via Headers in failure_route
by sadik.oualla.mohamed＠gmail.com 02 Jul '24

02 Jul '24

Hello, I am currently working on a scenario where I need to use Kamailio in the failure_route to process SIP responses, specifically handling negative SIP replies (e.g., 486 Busy Here) that Kamailio may receive. Here is the topology: ##B2BUA (acts as a UAC with the Kamailio, because it sends the INVITE) -> Kamailio -> Callee## In my setup, Kamailio needs to extract all Via headers from the received SIP response in the failure_route, focusing particularly on the bottom Via header (from the B2BUA' UAC part). However, I am encountering difficulties in achieving this. Here is what I have tried so far: 1. To extract the Via headers, I used the following line of code in the failure_route: ```$var(viaHeaderValues) = $T_rpl($hdr(Via));``` However, this only returns the topmost Via header. 2. I attempted to get the first and last Via headers values with: ```$var(viaHeaderValueFirst) = $T_rpl($hdr(Via)[0]); $var(viaHeaderValueLast) = $T_rpl($hdr(Via)[1]);``` Unfortunately, this also only returns the topmost Via header, which contains the address of Kamailio. My goal is to make sure that when Kamailio sends the SIP reply to the UAC with send_reply method on the failure_route, it uses the extracted *Via header* of the incoming SIP response (only the bottom one). Here is an example scenario: - Received SIP Response in kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.4;branch=z9hG4bK9157.090080a91105c0ec6279bb56882d1dc8.0 Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` - Desired SIP Response from kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` The current issue is that Kamailio modifies the Via header when sending the reply with send_reply method, resulting in: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m;received=200.200.200.3 From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` Due to this change, the UAC does not correctly react to or understand the SIP response "SIP Reply follows the RFC standards, but my current UAC waits for a Via that looks like Via of the previous SIP messages in the structure of parameters.". It is crucial for the Via header to maintain the same important order such as the branch, received, and rport values as the previous exchanged SIP messages. Could you please provide guidance on how to force Kamailio to send the SIP reply using send_reply method with the extracted bottom Via header from the incoming SIP response? Any advice or examples on handling this situation would be greatly appreciated. Thank you for your time and assistance.

1 0

Kamailio v5.7.6 Released
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, Kamailio SIP Server v5.7.6 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.6. For more details about version 5.7.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-7-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.7.x is currently the previous stable release series, the latest is 5.8.x and v5.8.2 was released on June 12, 2024. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

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Releasing v5.7.6
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, I am considering to release Kamailio v5.7.6 (out of branch 5.7) on Tuesday, July 2, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

SRV in advertise?
by David Villasmil 01 Jul '24

01 Jul '24

Hello guys, I'm trying to use an SRV as an advertised address but kamailio fails to start. Is this not possible? Thanks, David Villasmil email: david.villasmil.work(a)gmail.com

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SecSIPID Assistance
by Blake Ivey 01 Jul '24

01 Jul '24

Hi everyone. Wanting to see if someone could point me in the right direction. Still very knew to Kamailio but I am beginning to understand it better. I'm making an outbound proxy and have everything working well besides stir/shaken. I'm looking at the module page and have went back and forth with chatGPT and can't seem to figure this part out. I keep getting errors on the modparam lines. Obviously this is a self signed cert because I'm just testing. I am able to reach and download the cert from the Web server. Thank you for any assistance. # SECSIPID for Stir/Shaken modparam("secsipid", "private_key", "/etc/kamailio/secsipid/private.key") modparam("secsipid", "certificate", "/etc/kamailio/secsipid/cert.crt") modparam("secsipid", "authority_cert", "/etc/kamailio/secsipid/ca.crt") modparam("secsipid", "expire", 600) modparam("secsipid", "timeout", 2) route[STIRSHAKEN] { if (is_method("INVITE")) { if (!secsipid_add_identity("$fU", "$rU", "A", "", " http://myIPaddress.com/stir_shaken_cert.crt <http://myipaddress.com/stir_shaken_cert.crt>", "/etc/kamailio/secsipid/private.key")) { xlog("L_ERR", "Failed to sign call with ID: $ci - From: $fU\n"); send_reply("500", "Internal Server Error"); exit; } else { xlog("L_INFO", "Successfully signed call with ID: $ci - From: $fU\n"); } } # Relay the call after signing route(RELAY); }

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removing Via Header with received=
by mosbah abdelkader 30 Jun '24

30 Jun '24

Hello All, I am trying to build a simple proxy that forwards SIP requests to a remote portaone server. But, kamailio is adding a Via header with received=@IP to replies got from portaone. How to disable this behavior as Asterisk (client in my setup) is not accepting this header. Thank you.

1 0

misctest module not found in kamailio debian packages
by Patrick Karton 29 Jun '24

29 Jun '24

Hello, how do i get misctest module in debian packages ? it seems not included in a package. is compilation the only way to get it ? im using kamailio 5.7.5 on debian (bullseye) 11.8 version: kamailio 5.7.5 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 10.2.1 Thanks.

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Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org
by Xenofon Karamanos 28 Jun '24

28 Jun '24

Any news on regarding the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thanks, Xenofon ________________________________ From: Xenofon Karamanos via sr-dev <sr-dev(a)lists.kamailio.org> Sent: Wednesday, June 19, 2024 11:51 To: sr-dev(a)lists.kamailio.org Cc: Xenofon Karamanos <xk(a)gilawa.com> Subject: [sr-dev] Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org Hey everybody, While attempting to use the official Ubuntu package for Kamailio version 5.7.4 on Ubunut 24.04, we encountered an issue where TLS functionality does not seem to work as expected but 5.8 branch does. Given this, I am inquiring about the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thank you in advance for your assistance. Best regards, Xenofon

1 0

replies using the wrong protocol
by sarah.martin＠nextiva.com 27 Jun '24

27 Jun '24

I am extremely new at this, but trying to set up TLS with a carrier. TLS connection is good, Invite goes out, we get the 100 and the 200, but subsequent messages (ACK and BYE) are being sent with UDP and I cannot figure out how to get them to maintain the TLS transport. Any suggestions? I think this is the section I'm looking for. # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } if (has_body("application/sdp")) { xdbg("rtpengine_manage loop-protect MANAGE_REPLY"); #rtpengine_manage("loop-protect"); } }

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Kamailio cannot forward request - invalid sendtoparameters
by serkan celik 27 Jun '24

27 Jun '24

Hi all, I face a problem when trying to forward an UPDATE request to the IMS which communicates with UDP transport protocol on one side UDP and SCTP on other side. When I try to send UPDATE towards the UDP side of IMs I see following error on kamailio side and request cannot be forwarded Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: ERROR: <core> [core/udp_server.c:597]: udp_send(): sendto(sock, buf: 0x7fc180deb5f0, len: 1509, 0, dst: (172.18.0.20:5060), tolen: 16) - err: Invalid argument (22) Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: CRITICAL: <core> [core/udp_server.c:603]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net The UPDATE message which is tried to be forwarded is as follow: Here there are double rr one is UDP and one is sctp (Route: < sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>) ) That would be great if you can help why this UPDATE cannot be forwarded. By the way double rr is on on kamailio config Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: DEBUG: <core> [core/forward.c:577]: forward_request(): Sending:#012UPDATE sip:alice@10.10.141.188;ob;sbc-id=xpllrk47odqh4pb2ofpib6vsym SIP/2.0#015#012Via: SIP/2.0/UDP 127.0.0.1:50601;branch=z9hG4bKddab.3554948ce3cfe321729448a2b082c2f3.0#015#012Via: SIP/2.0/UDP 127.0.0.1:7060;branch=z9hG4bKnHN8Q3ya#015#012To: < sip:alice@ims.wingcon.com>;tag=49IZjY2irfb#015#012From: < sip:bob@ims.wingcon.com>;tag=hfdqjj2pmui#015#012Call-Id: 3mzzecWt3Cbgvy#015#012CSeq: 1 UPDATE#015#012Max-Forwards: 70#015#012Contact: <sip:bob@10.10.141.188;ob;sbc-id=o7dpxe7qu2r34zwzio5pis2vqi>#015#012Content-Type: application/sdp#015#012Route: <sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, <sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;transport=sctp;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.120;lr>#015#012Session-Expires: 180;refresher=uas#015#012Supported: replaces, 100rel, timer, norefersub#015#012Min-SE: 90#015#012P-Charging-Vector: icid-value=e277783432fd11efba03fa163eed425f;icid-generated-at=172.18.0.120;orig-ioi=172.18.0.120;term-ioi=172.18.0.120#015#012User-Agent: PJSUA v2.14 Linux-5.4.0.172/x86_64/glibc-2.35#015#012Content-Length: 409#015#012#015#012v=0#015#012o=- 3928313948 3928313950 IN IP4 172.18.0.120#015#012s=pjmedia#015#012b=AS:84#015#012t=0 0#015#012a=X-nat:0#015#012m=audio 30106 RTP/AVP 8 96 97 121 120#015#012c=IN IP4 172.18.0.120#015#012b=TIAS:64000#015#012a=ssrc:1986855169 cname:1fca52e812ced466#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96 AMR/8000#015#012a=rtpmap:97 AMR-WB/16000#015#012a=rtpmap:121 telephone-event/16000#015#012a=fmtp:121 0-16#015#012a=rtpmap:120 telephone-event/8000#015#012a=fmtp:120 0-16#015#012a=sendrecv#015#012a=rtcp:30107#015#012. Regards Serkan

1 0

Handling final response retransmissions after a reInvite
by hollis.rhonda＠gmail.com 26 Jun '24

26 Jun '24

I have a scenario where Kamailio is receiving a retransmit of a 200 OK to a late offer INVITE after it has sent the ACK, and after it has begun to handle a reInvite from the calling party. This results in the following commands to the rtpengine: Offer (reInvite), Offer (200 retransmit), Answer (ACK to retransmitted 200). At this point, I see STUN binding errors both in Chrome, where the webRTC client (called party) is running, and in the rtpengine logs. ERROR:port.cc(498)]: Received STUN BINDING error response: class=4 number=1 reason='Unauthorized' [core] STUN authentication mismatch from x.x.x.x:63396 [ice] Received ICE/STUN response code 487 for candidate pair dl2efuRG06eK4nFs:2713745946:1 from x.x.x.x::64509 to x.x.x.x [ice] ICE role change, now controlled [ice] Recalculating all ICE pair priorities [ice] Triggering check for dl2efuRG06eK4nFs:2713745946:1 [ice] ICE/STUN response with unknown transaction received (from x.x.x.x:64509 on interface x.x.x.x:16184) So, briefly, the call flow up to the point of error looks like this: INVITE w/o SDP 200 OK w/SDP, Offer to rtpengine ACK w/SDP, Answer to rtpengine reInvite w/SDP, Offer to rtpengine retransmission of 200 OK, Offer to rtpengine ACK to retransmitted 200, Answer to rtpengine I have dialog tracking enabled in my config, and it seems like Kamailio is behaving as expected when the retransmitted 200 and ACK are forwarded on, but I'm curious if other people have run into a scenario like this and how they have dealt with it.

1 0

Need help on how to implement presence_xml
by Michel Pelletier 25 Jun '24

25 Jun '24

Hello, I need to implement Event: presence, application/pidf+xml as described in RFC-3856. I understand I need to use the presence_xml module and generate PUBLISH requests to update the information and ensure that the NOTIFY get sent. I know nothing abour XCAP or wether I need it or not. Any information source out there that could help me? Regards, Michel Pelletier

1 0

TLS: client doesn't check certificate CN
by Marat Gareev 20 Jun '24

20 Jun '24

Hello, I noticed that Kamailio can route either to IP and FQDN. This means the server certificate CN is not checked by the client. How to enable something like the 'verify_peer' option? Thanks

1 0

Possibility to print values of Core Keywords in Kamailio configuration file
by sadik.oualla.mohamed＠gmail.com 20 Jun '24

20 Jun '24

Hello everyone, I have a question regarding the usage of core keywords, specifically the possibility of printing their values in log messages. While I understand that core keywords such as 'dst_port' and 'af' etc... can be accessed directly in if conditions, I am interested to know if there is a way to print the values of these core keywords for logging purposes. Thank you.

4 6

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Wed, 19 Jun 2024 11:54:05 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > What you should expect using this method is blocking while the exec runs > which could run you out of child processes while they complete. I'd also > expect to see an "unusual amount of CPU activity" which will be the result > of the forked shells being created and destroyed. > > All of that being said, yes it works. It's architecturally overly simple > and very inefficient. If your load is tiny, it might be ok. Anything > moderate like Henning and others have said, will use an inordinate amount > of cpu. I would not personally risk it. > > Thank you Brett, Henning and Alex Sadly the load won't be tiny and cannot risk it. For now I'll deploy a local nginx and fpm to run the scripts, which are in php, and will use the http_async method. Sadly we have to "http all the things" nowadays. Let's see if I'm able to write down all the logic tomorrow and do some testing. cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

Adding custom headers to HTTP server response xhttp
by Sasivarunan 19 Jun '24

19 Jun '24

HI Team, I use tls and xhttp module to enable websocket interface for webrtc client to connect. I wanted to add "Strict-Transport-Security" header to xhttp response to add security. But currently I don't see a way to add it anywhere. I can see only server_header as an option, But this will get added to SIP messages as well. It 'd be great if anybody has a suggestion around this. Thanks Varunan

2 2

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Tue, 18 Jun 2024 13:54:41 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > Just want to add that exec is heavy and slow. I would not recommend it. > > Is there a reason you want to do this over http-ifying your script and > using async? > > Not really. The scripts are provided. Maybe I'm against "http all the things" which seems to be what everyone does nowadays. Why would http client be lighter than executing a script? That would be a design problem, right? -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

exec concurrency
by Jon Bonilla (Manwe) 18 Jun '24

18 Jun '24

Hi all I've been working with async http client and async db queries in the past but now I have to execute scripts and store the return values in avps. Since I have no experience with that, I wonder how to achieve concurrency with that scenario. Is there a way to exec async or suspend until the exec has finished and resume later? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 2

Psresence dialoginfo question
by Stefan Mititelu 17 Jun '24

17 Jun '24

Hello, I am doing BLF but I want different users to be able to setup different dialog info initial states. My question is if there's an already implemented way for doing that? I've looked over docs and C code so far and this is what I've found: 1. modparam("presence_dialoginfo", "force_dummy_dialog", 1) => will force sending initial NOTIFY with dialog state "terminated" 2. Function calls: dlginfo_agg_nbody()->dlginfo_agg_nbody_empty() which will build the pre-defined xml body with that "terminated" state. If not implemented so far, maybe this can be used to set a different initial state for a user (e.g. export a cfg function or take that info from DB?) Any opinions on this? Thank you, Stefan

2 2

db_mysql startup warnings
by Juha Heinanen 17 Jun '24

17 Jun '24

When I start Kamailio master version that has been build on Debian 12 with default-libmysqlclient-dev depending on libmariadb-dev-compat, I get several warnings like this: 2024-06-17T14:48:27.702827+03:00 siika /usr/bin/sip-proxy[297246]: WARNING: db_mysql [km_my_con.c:187]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring How can I prevent them? -- Juha

1 1

How to Check Source Port in Application Layer with Kamailio
by sadik.oualla.mohamed＠gmail.com 16 Jun '24

16 Jun '24

Hello kamailio lovers, I am currently working on a Kamailio configuration where I need to check the source port of incoming SIP messages. Initially, I used the $sp pseudo-variable to get the source port, which worked perfectly while testing with SIPp. Here is the snippets code that I use " if ($sp != 5080 && $sp != 8080 && $sp != 5060){ sl_send_reply(403, "forbidden"); xlog("this $rm received, is forbidden\n"); exit; } " However, I encountered an issue when I switched to using sipsak for testing. It seems that sipsak uses a randomly assigned port for the actual network connection, even though it sets the source port in the SIP headers (e.g., Via, To and Contact headers) to a specified port using the -l option. Here is the sipsak command that I use "sipsak -vv -s sip:jack@192.168.1.2:5060-l 5080" My goal is to have Kamailio check the source port as specified in the application layer, rather than the transport layer port (i.e., the ephemeral port assigned by the OS randomly like 54691, 33017...). Could anyone guide me on how to properly configure Kamailio to extract and check the port from the application layer or SIP headers, or am I doing this in the wrong way? Any advice or examples would be greatly appreciated. "I am not a native english speak, so I use chatGPT to make my questions clear" Thank you!

1 0

topos module issue on hangup
by Arun K R 13 Jun '24

13 Jun '24

Hello, I am using topos module to hide Record-Route and Route header as per the requirement from carrier. But the topos module hides, contact, via and routing headers also. Due to this reason when I am getting bye from the carrier I am not able to route the bye back to the agent. Bye from Agent to carrier is working fine. Bye from carrier to my internal network interface is not routing to client through public interface. I assume this may be because Via / Contact header has been hidden and changed by the topos module with the invite please help me for this Thanks Arun

3 5

Kamailio v5.8.2 Released
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, Kamailio SIP Server v5.8.2 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.2. For more details about version 5.8.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/06/kamailio-v5-8-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v5.8.2
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, I am considering to release Kamailio v5.8.2 (out of branch 5.8) on Wednesday, June 12, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 2

New features in v.5.8
by Leonid Fainshtein 10 Jun '24

10 Jun '24

Hi, In the "Kamailio v5.8.0 Release Notes" I see: "SIP overload control support for server (RFC7339)". Where can I find some more details about the feature? Also, the similar question about the new "influxdbc" module. What are the possible use cases for this module? Thank you, Leonid Fainshtein Xorcom Ltd

2 2

handling 300 redirect requests
by Rob Green 09 Jun '24

09 Jun '24

I am trying to proxy a request from server A to server C with me being server B. i send the invite from A to B, then i need to send to server R which returns back to me a 300 redirect, which i then need to take the answer from the Contsct and then proxy the original invite from A to C. server A can not handle 3xx responses and must be proxied. I havent been able to find any examples where the 300 is intercepted and a new invite to the target is sent. i have the following code but it results in no new INVITE and the original session is hung until it is canceled. Any idea what i may be doing wrong? In the example below i am just hardcoding the uri for the new invite instead of getting from the Contact as a debug exercise. I am using Kamailio 5.7.3 route[HANDLEREDIRECT] { xlog("L_INFO", "DBG: HANDLEREDIRECT"); msg_apply_changes(); $var(test) = $hdr(Contact); xlog("L_INFO","DBG: redirect headers [$var(test)]: from=[$fu] from_user=[$fU], to=[$tu] to_user=[$tU]"); # send to gateway $var(new_uri) = "sip:" + $tU + "@" + "gateway"; if ($var(new_uri) != "") { xlog("DBG: Redirecting to: $var(new_uri)\n"); $ru = $var(new_uri); $tu = $ru; record_route(); route(RELAY); exit; # i have also tried using #t_newtran(); # and #t_relay(); } else { xlog("DBG: No contact header found in 300 response\n"); } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("DBG: new branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "DBG: manage branch"); route(NATMANAGE); return; } onreply_route[MANAGE_REPLY] { xdbg("DBG: incoming reply\n"); xlog("L_INFO", "DBG: manage reply"); if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: got 3xx redirect"); route(HANDLEREDIRECT); drop; } return; } I also tried using uac_redirect module, but it stalls the incoming invite also, and i cant figure out how to modify the uri from the contact before sending (to add params) # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: following 3xx redirect"); get_redirects("3:1"); t_relay(); exit; } } Rob Green Chief Information Officer (CIO) Tel: 1-352-218-1490 Cell: 1-407-620-0932 Email: rob.green(a)nuwave.com Web: www.nuwave.com This communication is the Property of NUWAVE Communications and may contain confidential or privileged information. Unauthorized use of this communication is strictly prohibited and may be unlawful. If you have received this communication in Error, Please immediately notify the sender by reply e-mail and destroy all copies of the communication and any attachments.

2 2

Setting Key/Value Pairs at Dialogue Level
by Pavan Kumar 08 Jun '24

08 Jun '24

Hi everyone, I am creating a dialogue when I receive INVITE. Is there a way to set key/value pair like attributes at dialog level, so that, when I receive BYE message I can retrieve that information and do some processing? Thank you, Pavan Kumar

2 2

Recommendation for a signaling B2BUA to work with Kamailio on Ubuntu
by Benoît Panizzon 07 Jun '24

07 Jun '24

Hi I wonder if anyone could maybe recommend what to use as signaling B2BUA to work with Kamailio to handle spiralling / looping calls and to reduce the number of Route, Via Header and message size, which cause issue with a large number of CPE. Requirements: * New CallID needs to be generated * Handling Route and Via * ~ 2000 concurrent calls. * Preserving custom SIP X- header * Routing NOTIFY for MWI etc. Handling RTP is not required! What I have been looking at more closely: = FreeSwitch = Got stuck trying to solve the dependencies to get it to run/build on ubuntu jammy. Has anyone experience in running it in a docker container? Will it perform? = Sippy = Very small, very simple, unfortunately removes customer SIP headers and does not forward NOTIFY. (It's python, so maybe this could be easily implemented/fixed). = SEMS = Does not seem to be actively developed in the last, couple of years. Looks quite bloated to me for what I need. = Asterisk = Each customer header has to be copied, wants to handle RTP and bridge by issuing Re-Invites. Not sure how to route stuff like NOTIFY. Also from experience I fear asterisk would translate sip reply codes (like 608 and 607) it does not know to some 4XX default. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 3

Filtering codes in db acc
by Jon Bonilla (Manwe) 06 Jun '24

06 Jun '24

Hi all I'm trying to make kamailio not to insert a row in missed_calls table when reply code is 407. But even if I set the "failed_filter" modparam the 407 row is inserted. The acc mmodparams are: modparam("acc", "db_table_acc", "kam_acc") modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACC_MISS) modparam("acc", "db_url",DBURL) modparam("acc", "db_insert_mode", 1) modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "failed_filter", "401,407") modparam("acc", "db_extra", "whatever" db is mysql. Kamailio version is 5.7.4 When I send a call that receives a 407 and after auth a 480 y get both rows in missed_calls table. Log: DEBUG: acc [acc_mod.c:404]: mod_init(): acc db table initialized to: kam_acc DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 0 = 401 DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 1 = 407 DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cec807480) event type 512, reply code 407 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 407 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cebf59bc0) event type 512, reply code 486 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 486 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off am I configuring this wrong? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 3

CentOS 7 and 8 deprecated
by Sergey Safarov 06 Jun '24

06 Jun '24

The CentOS project page shows that the 7 and 8 releases are deprecated. https://www.centos.org/ Does the Kamailio community use these releases for prod? If yes then I will keep the packaging for this release. If not, I will remove the packaging for 7 and 8 releases, and in the future Kamailio will be packaged for 9 and 10 releases (planned). https://www.phoronix.com/news/CentOS-Stream-10-Start https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre… https://composes.stream.centos.org/stream-10/development/latest-CentOS-Stre…

2 1

Assistance Needed: Converting 11-Digit US Numbers to E.164 in Kamailio
by Pavan Kumar 04 Jun '24

04 Jun '24

Hey everyone, I have the following setup: External User/Trunk <===> Kamailio <===> FreeSWITCH I have configured FreeSWITCH and Kamailio to work only with phone numbers in E.164 format. Recently, I needed to integrate a new SIP trunk that sends phone numbers in an 11-digit US format. My initial idea to solve this is to use Kamailio as a translator that converts 11-digit numbers to E.164 when sending to FreeSWITCH, and back to 11-digit when sending to the trunk. Before writing the code, I want to know if there is already a solution to this problem in one of the modules, so that I won't reinvent the wheel. Thanks, Pavan Kumar

7 11

DMQ_USRLOC and multiple domains
by Lợi Đặng 03 Jun '24

03 Jun '24

Hi, I'm setting up shared user location for 2 instances of kamailio via dmq_usrloc. Everything seems fine except my setup is multi-domains. The issue arises when the producer node produces the Contact of domain X, Contact of domain Y separately, but the consumer node stores both as the Contact of domain Y (seen in kamcmd ul.dump), in which Y is set in usrloc_domain. modparam("dmq_usrloc", "usrloc_domain", "{{domain_Y}}") modparam("usrloc", "use_domain", 1) ==== I had a quick ngrep and see the domain was put in the "aor" key of the body in KMQ message for usrloc {"action":1,"aor":"test_username_0dppxcl@$domain_X"} But I'm not sure how to make my consumer node to store the contact in appropriate domain. Any help will be appreciated. Thanks and regards, Loi Dang Thanh Phone : +84. 774.735.448 Email : loi.dangthanh(a)gmail.com

1 1

ACK and Bye not relaying when record_route is not used
by Arun K R 03 Jun '24

03 Jun '24

Issue : Not getting relay of ACK and BYE to the next hop after the call is answered my Scenario : Asterisk ------->kamailio sip proxy-------------------> carrier (outgoing call) My carrier is not allowed to send the SIP packet with Record-Route header. So that I have removed record_route(). After that the call is getting connected. I am getting 200 OK (SDP) from carrier side and forwarded that to the Asterisk on the other side. As a response I am getting ACK from asterisk. But the kamailio is not forwarding the ACK to the carrier side. I understood this is because the record-route is not there. The same thing is happening for BYE also. The Bye is not forwarding to carrier side. Kindly suggest me a solution for this for relaying ACK and bye without Record-Route in kamailio Bellow is the 200 OK SDP I am sending back to asterisk 2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060 SIP/2.0 200 OK Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 Via: SIP/2.0/UDP 103.182.153.113:5060 ;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 CSeq: 22823 INVITE Allow: INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE Contact: <sip:lucentNGFS-115804@10.5.110.117:5060 ;alias=10.5.110.117~5060~1;x-afi=11> Content-Type: application/sdp Session-Expires: 7200;refresher=uas Supported: timer Content-Length: 248 v=0 o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101 s=- c=IN IP4 103.155.114.101 t=0 0 m=audio 12806 RTP/AVP 8 101 ------------------------------------------------------------------------------------------------------- The ACK I am getting back from asterisk is 2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060 ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11 SIP/2.0 Via: SIP/2.0/UDP 103.182.153.113:5060 ;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce From: <sip:917946357720@gaesip.teleforce.in >;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1 To: <sip:09496381412@103.155.114.101 >;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109 Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2 CSeq: 22823 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 18.13.0 Content-Length: 0 Thanks Arun

3 3

StriShaken module compilation
by Sergio Charrua 01 Jun '24

01 Jun '24

Hello all! I am struggling on compiling StriShaken module on RHEL 9.2. So far, the module was compiled as well as libstirshaken. Below are the steps used to compile (as sudo) both library and module (I hope this may help someone) and, of course, correct me if I am wrong in any step or if there is a better way. Also, note that the RHEL 9,2 is in a VM environment without access to the WWW, instead it uses Red Hat Satellite to download packages (via Yum or DNF), if available. Lib LibKS download libks from https://github.com/signalwire/libks unzip move to dir created by Unzip run yum groupinstall "Development Tools" dnf install libuuid-devel libatomic openssl-devel cmake . make make install cp /usr/lib/pkgconfig/libks2.pc /usr/lib64/pkgconfig/. For module LibStirShaken: download https://github.com/signalwire/libstirshaken unzip move to dir if using OpenSSL3.0 or + edit file configure.ac and add after line 28: if test x$HAVE_OPENSSL = x1; then openssl_CFLAGS="$openssl_CFLAGS -DHAVE_OPENSSL"; /* PATCH FOR OPENSSL3 */ AC_MSG_CHECKING([for OpenSSL >= 3.0.0]) AC_COMPILE_IFELSE([AC_LANG_PROGRAM([[ #include <openssl/opensslv.h> #if OPENSSL_VERSION_PREREQ(3,0) #error "you_have_version_3" #endif ]], [[]])], [ AC_MSG_RESULT([no]) ], [ AC_MSG_RESULT([yes]); AC_DEFINE(OPENSSL_SUPPRESS_DEPRECATED, 1, [disable openssl deprecated-function warnings]) ]) /* END OF PATCH */ else AC_MSG_ERROR([OpenSSL >= 1.0.1e and associated developement headers required]) fi run configure make sudo make install next, move to dir /root/kamailio/kamailio-5.7.4/src/modules/stirshaken cd /root/kamailio/kamailio-5.7.4/src/modules/stirshaken ln -s /root/stirshakenlib/libks-master/src/include/libks libks make make install All seem to compile OK , *but* when starting Kamailio I get the following errors: May 29 17:42:02 kamailio1 kamailio[248991]: ERROR: <core> [core/sr_module.c:599]: ksr_load_module(): could not open module </usr/local/lib64/kamailio/modules/stirshaken.so>: libstirshaken.so.1: cannot open shared object file: No such file or directory May 29 17:42:02 kamailio1 kamailio[248991]: CRITICAL: <core> [core/cfg.y:3915]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio_stir_shaken_mod.cfg, line 106, column 12-26: failed to load module I haven't found a solution for this. Can anyone help? Thanks in advance! *Sérgio Charrua*

2 1

Parsing Error with SIP 607 Response
by Ghulam Mustafa 31 May '24

31 May '24

Hello Kamailio Community, I'm encountering a parsing error in Kamailio when receiving a SIP 607 "Unwanted" response. The relevant log entries are: May 31 12:32:09 k-proxy1 /usr/sbin/kamailio[320898]: ERROR: <core> [core/receive.c:378]: receive_msg(): core parsing of SIP message failed (36.53.xx.xx:5060/1) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/parse_fline.c:272]: parse_first_line(): parse_first_line: bad message (offset: 11) May 31 12:32:10 k-proxy1 /usr/sbin/kamailio[320895]: ERROR: <core> [core/parser/msg_parser.c:790]: parse_msg(): ERROR: parse_msg: message=<SIP/2.0 607#015#012Via: SIP/2.0/UDP 134.195.195.40:5060;branch=z9hG4bKc87a.ed6403ae1529f36b65171bf964604be0.0;i=2#015#012Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFIzl5c9wbfNG3oaHkl3#015#012From: "15516756486" <sip:15516756486@10.x.x.10>;tag=KXtSBZ1at71Hc#015#012To: <sip:99904441743709685@36.53.xx.xx;transport=udp>;tag=sbcsipuas_1_C35684_20240531083209909_sbc06#015#012Call-ID: a40d965d-99ec-123d-6f92-ecf4bbd64f6c#015#012Contact: <sip:36.53.xx.xx:5060>#015#012CSeq: 84025572 INVITE#015#012Server: sbc_5#015#012Content-Length: 0#015#012#015#012> And the actual SIP message is U 36.53.xx.xx:5060 -> 10.x.x.30:5060 #850 SIP/2.0 607. Via: SIP/2.0/UDP 134.195.195.40:5060 ;branch=z9hG4bKbdbd.95060a558d66b77d258b8d0799be4cf0.0;i=2. Via: SIP/2.0/UDP 127.0.0.8 ;branch=z9hG4bKsr-Sz6QPbTDsrqSQc7ys87Ds87VP37Ds87YG870sXCWo1G6ujo6oXzisr0isX7Dsr0isXCSZXcFujwtH.KOH.QqGXs09XNtw.dEJ5GhvjhauNHzwFCaHXyLA3G.SjwaAmdn. From: "13xxxxxxxxxx" <sip:13xxxxxxxxxx@10.x.x.10>;tag=pFFatBpUc75BH. To: <sip:441xxxxxxxxx@36.53.xx.xx ;transport=udp>;tag=sbcsipuas_1_C56639_20240531081932874_sbc04. Call-ID: e0d2f5fa-99ea-123d-6f92-ecf4bbd64f6c. Contact: <sip:36.53.xx.xx:5060>. CSeq: 84025194 INVITE. Server: sbc_5. Content-Length: 0. Any insights or suggestions would be greatly appreciated. Thank you! Mustafa

5 9

Access avp vars in a cancel? How to pass information from an INVITE to a CANCEL?
by Benoit Panizzon 31 May '24

31 May '24

Hi To determine, if rtpengine was engaged during an invite or not, I make use of rr params for all messages in following transactions and I set an avp for all messages in the current transaction. When the call is canceled, I would need to call rtpengine_delete to make sure rtpengine stops listening for rtp packets. Unfortunately, the avp I set during the INVITE, does not seem to be present while the CANCEL message is processed, despite to my understanding being in the same transaction. How can I pass some information from an INVITE to the corresponding CANCEL? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 10

core dump on AWS with ~4000 TCP sockets
by David Villasmil 31 May '24

31 May '24

Hello all, I'm seeing core dumps (can't find them!) on AWS with lots of TCP sockets open (not related to load), kamailio dies randomly with: *May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743739]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 142 May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743542]: ALERT: <core> [main.c:788]: handle_sigs(): child process 743672 exited by a signal 11 May 30 14:31:03 sip-proxy /usr/sbin/kamailio[743542]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated May 30 14:32:03 sip-proxy /usr/sbin/kamailio[743542]: CRITICAL: <core> [main.c:698]: sig_alarm_abort(): shutdown timeout triggered, dying...* before this happens, there's a ton of May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.56.51.197:34846 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 98.196.165.187:38593 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:49372 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 97.101.162.40:45248 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 172.59.213.36:57074 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 207.195.63.14:2521 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:22155 failed (timeout) May 26 00:32:01 sip-proxy /usr/sbin/kamailio[74966]: ERROR: <core> [core/tcp_main.c:4591]: tcpconn_main_timeout(): connect 10.241.49.185:10339 failed (timeout) Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

Re: Via header force change protocol to TLS?
by David Villasmil 31 May '24

31 May '24

trying to use msg_apply_changes(); subst_hf("Via","/TCP/TLS/","f"); But as i don't seem to have access the top-most via, i can'it doesn't work... i cam change the incoming Via, but not the outgoing... I need this urgently.. any help anyone? > Sergey, > > Thanks for pointing me to the PR but that's not exactly what i need, let > me explain: > > We have an AWS an NLB on which our clients connect to and terminate TLS > connections and from then on into our proxies it is TCP. Now when clients > send invites (with TLS in their VIA as protocol) and we reply to them > (because the socket is TCP) I need to set the protocol on the Via to be > TLS. Otherwise, the client would not understand. This message or will > believe that it's not. about the same connection > > I haven't figured out a way of doing this since it can't get the actual > VIA that will be sent out. I've tried on the onsend_route (enabling > onsend_route_reply) > > Help is greatly appreciated! > > > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > > > On Thu, May 30, 2024 at 11:44 PM David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> HEllo Sergey, >> >> i can send one. yes. >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work(a)gmail.com >> phone: +34669448337 >> >> >> On Thu, May 23, 2024 at 5:14 PM Sergey Safarov <s.safarov(a)gmail.com> >> wrote: >> >>> Hi David >>> Could you send PCAP for an inbound call via TCP connection? >>> >>> Sergey >>> >>> On Thu, May 23, 2024 at 5:53 PM David Villasmil < >>> david.villasmil.work(a)gmail.com> wrote: >>> >>>> it's still in progress though. >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work(a)gmail.com >>>> phone: +34669448337 >>>> >>>> >>>> On Thu, May 23, 2024 at 4:51 PM David Villasmil < >>>> david.villasmil.work(a)gmail.com> wrote: >>>> >>>>> Thanks, I'll check it out! >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work(a)gmail.com >>>>> phone: +34669448337 >>>>> >>>>> >>>>> On Thu, May 23, 2024 at 4:16 PM Sergey Safarov <s.safarov(a)gmail.com> >>>>> wrote: >>>>> >>>>>> We have tested this PR using the Linphone app. >>>>>> So your case will be resolved using this PR. >>>>>> Need to enable HAproxy protocol headers. >>>>>> >>>>>> On Wed, May 22, 2024 at 4:36 PM Sergey Safarov <s.safarov(a)gmail.com> >>>>>> wrote: >>>>>> >>>>>>> Please try Kamailio PR >>>>>>> https://github.com/kamailio/kamailio/pull/3731 >>>>>>> >>>>>>> We have developed this PR for use case you have described. >>>>>>> We have tested Route and Record-Route headers not Via. >>>>>>> So will provide some review for this PR then will be fine. >>>>>>> >>>>>>> On Wed, May 22, 2024 at 2:22 PM David Villasmil < >>>>>>> david.villasmil.work(a)gmail.com> wrote: >>>>>>> >>>>>>>> Hello Sergey, >>>>>>>> >>>>>>>> Thanks for the suggestion. Not sure if his is what i'm looking for, >>>>>>>> allow me to explain further: >>>>>>>> We set up an NetworkLoadBalancer on AWS to offload tls on it. This >>>>>>>> Load balancer is a TLS listener on the outside and a TCP connection to the >>>>>>>> proxy inside. >>>>>>>> So when sending an INVITE to the connected client, the via has a >>>>>>>> TCP protocol like >>>>>>>> >>>>>>>> Via: SIP/2.0/TCP >>>>>>>> mydomain:port;branch=z9hG4bKf176.53ac8af0d7090a31e44548f15ea420ff.0 >>>>>>>> >>>>>>>> and the client (linphone) disconnects and tries to contact the >>>>>>>> proxy on that address on a TCP socket, which doesn't exist. I tried many >>>>>>>> solutions none of which actually work... last one setting $du =$du + >>>>>>>> ";transport=tls" and forcing the socket to the TCP socket to the load >>>>>>>> balancer, but of course i'm getting warnings about this. >>>>>>>> >>>>>>>> is this something that PR (not merged) would be addressing, i >>>>>>>> didn't see that. >>>>>>>> If not, is there a way of doing this without any trickery? >>>>>>>> >>>>>>>> Thanks! >>>>>>>> >>>>>>>> Regards, >>>>>>>> >>>>>>>> David Villasmil >>>>>>>> email: david.villasmil.work(a)gmail.com >>>>>>>> phone: +34669448337 >>>>>>>> >>>>>>>> >>>>>>>> On Wed, May 22, 2024 at 12:16 PM Sergey Safarov < >>>>>>>> s.safarov(a)gmail.com> wrote: >>>>>>>> >>>>>>>>> Probable you need this PR >>>>>>>>> https://github.com/kamailio/kamailio/pull/3810 >>>>>>>>> >>>>>>>>> Or you can try >>>>>>>>> https://github.com/kamailio/kamailio/pull/3731 >>>>>>>>> In this PR we faced the same issue and solved this. >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, May 22, 2024 at 3:43 AM David Villasmil via sr-users < >>>>>>>>> sr-users(a)lists.kamailio.org> wrote: >>>>>>>>> >>>>>>>>>> Hello Anthony, did you solve this problem? I'm facing the same >>>>>>>>>> problem >>>>>>>>>> >>>>>>>>>> Thanks! >>>>>>>>>> Regards, >>>>>>>>>> >>>>>>>>>> David Villasmil >>>>>>>>>> email: david.villasmil.work(a)gmail.com >>>>>>>>>> phone: +34669448337 >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> On Mon, Feb 5, 2018 at 5:57 AM Anthony Alba < >>>>>>>>>> ascanio.alba7(a)gmail.com> wrote: >>>>>>>>>> >>>>>>>>>>> I have kamailio behind a TLS termination proxy so the sockets >>>>>>>>>>> are correctly deduced to be TCP. However the clients only talk TLS to the >>>>>>>>>>> proxy and are confused when the top Via header added by Kamailio is TCP. Is >>>>>>>>>>> there a way for Kamailio to forcibly pretend its protocol is TLS? Like >>>>>>>>>>> advertised_address but "advertised_protocol" instead. >>>>>>>>>>> >>>>>>>>>>> (With pjsip testing: it has a flag use_tls which ignores TCP >>>>>>>>>>> from Kamailio and continues to use the persistent TLS transport to proxy. >>>>>>>>>>> Linphone fails because it tries to honor TCP in Via and is unable to >>>>>>>>>>> establish TCP transport). >>>>>>>>>>> >>>>>>>>>>> BTW I am using t_relay_to_tcp so Kamailio will return traffic to >>>>>>>>>>> the proxy as TCP even though the contact addresses specify transport=TLS. >>>>>>>>>>> _______________________________________________ >>>>>>>>>>> Kamailio (SER) - Users Mailing List >>>>>>>>>>> sr-users(a)lists.kamailio.org >>>>>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>>>>>>>>> >>>>>>>>>> __________________________________________________________ >>>>>>>>>> Kamailio - Users Mailing List - Non Commercial Discussions >>>>>>>>>> To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >>>>>>>>>> Important: keep the mailing list in the recipients, do not reply >>>>>>>>>> only to the sender! >>>>>>>>>> Edit mailing list options or unsubscribe: >>>>>>>>>> >>>>>>>>>

1 0

Inbound Identity Verification Stir/Shaken
by tfayomi＠untangledtechnology.com 30 May '24

30 May '24

Hi All, I have a quick question regarding verification on incoming calls regarding the stir/shaken policy. I'm using the secsipid library to check the identity header, but the problem I'm running into is that there is no identity header in the invite from the incoming call (It's showing up as null). Have most carriers enabled passing along an identity header, or is this something that is still in the works for some of them. Would I need to request and get the certificate from the terminating carrier? I've tried using the auth_identity module to see if i need to use the verification functions to get/request the certificate, but I have had no luck so far. Would anyone be able to provide direction on how to request/get/pass along the identity header from an incoming call with stir/shaken. (I am using kamailio v5.6 and debian 12) Thanks in advance, Temi

4 6

Via header force change protocol to TLS?
by Anthony Alba 30 May '24

30 May '24

I have kamailio behind a TLS termination proxy so the sockets are correctly deduced to be TCP. However the clients only talk TLS to the proxy and are confused when the top Via header added by Kamailio is TCP. Is there a way for Kamailio to forcibly pretend its protocol is TLS? Like advertised_address but "advertised_protocol" instead. (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio and continues to use the persistent TLS transport to proxy. Linphone fails because it tries to honor TCP in Via and is unable to establish TCP transport). BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy as TCP even though the contact addresses specify transport=TLS.

4 10

libstirshaken installation issues
by tfayomi＠untangledtechnology.com 30 May '24

30 May '24

Hi All, I am attempting to install the stirshaken module for kamailio and I ran into a couple issues installing the open source c library for libstirshaken. When I use the make command, I'm getting this error: src/stir_shaken.c: In function 'stir_shaken_is_key_trusted': src/stir_shaken.c:726:9: error: 'EVP_PKEY_cmp' is deprecated: Since OpenSSL 3.0 [-Werror=deprecated-declarations] 726 | if (!EVP_PKEY_cmp(pkey, candidate_pkey)) { | ^~ In file included from /usr/include/openssl/x509.h:29, from /usr/include/openssl/ssl.h:31, from /usr/include/libks2/libks/ks_ssl.h:25, from /usr/include/libks2/libks/ks.h:80, from include/stir_shaken.h:15, from src/stir_shaken.c:1: /usr/include/openssl/evp.h:1418:5: note: declared here 1418 | int EVP_PKEY_cmp(const EVP_PKEY *a, const EVP_PKEY *b); | ^~~~~~~~~~~~ At top level: cc1: note: unrecognized command-line option '-Wno-gnu-zero-variadic-macro-arguments' may have been intended to silence earlier diagnostics cc1: all warnings being treated as errors make: *** [Makefile:1337: src/stir_shaken.lo] Error 1 I tried going into the stir_shaken file and changing EVP_PKEY_cmp to EVP_PKEY_eq, but it just brings up more deprecation errors. I also tried rolling back the openssl version on my machine to openssl version 1.1 and changing EVP_PKEY_eq back to EVP_PKEY_cmp, but the same error persists. Is there any way to resolve this error, or is there another library that can be used to get the stirshaken module for kamailio? Thanks in advance, Temi

6 7

Is udp_mtu and udp_mtu_try_proto broken in kamailio v5.8
by Supreeth Herle 30 May '24

30 May '24

Hello, I have a dockerized IMS setup using kamailio from git commit id 4fb8accc6747ad56fec3dc84d70cb2b8bbd7316e (tag 5.8.x), where I am seeing an issue faced by one of the users of the dockerized IMS. The issue is that the SIP request (in UDP) even though it exceed the MTU of 1300 (udp_mtu=1300 and udp_mtu_try_proto=TCP) the SIP request from P-CSCF is not sent out as a TCP. The pcap and P-CSCF logs in debug can be found in the github issue - https://github.com/herlesupreeth/docker_open5gs/issues/316#issuecomment-213… IP of the containers are follows: P-CSCF - 172.22.0.21 S-CSCF - 172.22.0.20 SIP Client - 192.168.101.x P-CSCF is configured with udp_mtu=1300 and udp_mtu_try_proto=TCP In the pcap you can notice that the SIP NOTIFY originating from S-CSCF is forwarded to P-CSCF (packet no. 698) and packet length is 2716. At this point P-CSCF should have tried to convey the SIP packet to SIP client via TCP. But I dont see any TCP connection establishment either and eventually request times out (packet 1147). [image: image.png] Any help or insights would be highly appreciated. Thanks and Regards, Supreeth

1 0

Re: Failover with dialog dmq while dialog state 2
by Björn Klasen 27 May '24

27 May '24

Hi Henning, thank you for your reply. I did a lot testings during the past days and I think I found a solution that is working, although it's very dirty... I found out it is possible to send a manual KDMQ-Message to all Proxy-Nodes including the one that it sending die KDMQ-Message. So I create a dialog state 4 message on ACK and indeed the Dialog on all Peers are Updated correctly to state 4. I also tried this for state 3 on 200 OK, but this does not work. I haven't had a look into the code yet but my experiments shows, that there is never is state 3 update via DMQ, so I think it's not implemented. Interestingly the TM-Timer still times out after 300 seconds but in normal circumstances this doesn't bother, because either the failed Kamailio will not trigger this, because the Software or the Server crashed, or it tries to send a CANCEL that is denied because we are in dialog state 4. The only thing is your really have to be careful about your hash tables. You have to delete them at the correct point. I really need to test a lot at this point, but till now, billing seems to be OK even on failover scenarios. This is why I really like Kamailio :) BR, Björn Am 09.05.24 um 20:57 schrieb Henning Westerholt: Hello, thanks for the detailed e-mail. As also indicated in the module documentation, the dialog module DMQ replication will not replicate everything, its main use-case is for profile data sharing. https://kamailio.org/docs/modules/5.8.x/modules/dialog.html#dialog.p.enable… In the past months there have been some other discussions on the users lists about similar scenarios (I think related to billing/accounting) and dialog with DMQ, which might be interesting for you in this regard. If you find issues where the DMQ synchronisation is lacking some functionality in the dialog module, you can create a feature request in our issue tracker. There is of course no guarantee that this limitation is also timely addressed. Regarding the INVITE and CANCEL scenario, this is usually not related to dialog but to the tm module. As you also mentioned, there is no replication of transaction state in tm. Cheers, Henning -- Björn Klasen, Senior Specialist (VoIP) TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de<mailto:bklasen@tng.de> https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Gunnar Peter, Sven Schade, Carsten Tolkmit, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

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I'm looking for an get_rr_param() usage example
by Benoit Panizzon 27 May '24

27 May '24

Hi Gang To process spiraling calls without the dialog module. I'm trying to use a rr param to identify in which iteration I am. So when I have a call spiraling through the same instance 3 times, every time I process an invite I would do something like: add_rr_param(";sp-count=$avp(sp-count)"); But to initialize that counter, I need to be able to read what is in there beforehand. if (check_route_param("sp-count=") { get_route_param(msg, "sp-count", $avp(sp-count)) $avp(sp-count) = $avp(sp-count) + 1; add_rr_param(";sp-count=$avp(sp-count)"); } else { add_rr_param(";sp-count=1"); } But what do I pass as msg parameter? Is it the PV $msg(hdrs) ? PS: I need this to append this to the call ID for rtpengine so I hopefully can match new transactions no matter from which side they are initiated to the correct rtp stream. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 0

Is there any funtion to route based on source phone number
by SAMUEL MOYA TINOCO 27 May '24

27 May '24

Hi everyone, I have a kamailio configured to send invites from different phone numbers registered in kamailio to different endpoints. The configuration that I have right now is a big if chain with the phone numbers that then sends the invite using dispatcher module. Example: If (phone number in invite = phone number X) { dispatcher sends to endpoint X} If (phone number in invite = phone number Y) { dispatcher sends to endpoint Y} If (phone number in invite = phone number Z) { dispatcher sends to endpoint Z} Im not sure if this is the correct way of doing this or if there is any module that can do this allowing me to configure it using BBDD instead of kamailio configuration. I started doing it like this on our test environment while I was learning how to use kamailio and I'd like to know if this is the right way to do it or if I should do it with other methods Thank you in advance

7 7

Fwd: rtpengine timestamp jumps
by Jeff Brower 26 May '24

26 May '24

Rtpengine support- I am re-posting again, as I noticed on the SR-USERs web page no HTML or inline images are showing (https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…). The Wireshark screencap that demonstrates the issue is uploaded at: https://www.signalogic.com/images/rtpengine_wireshark_capture_timestamp_jum… all other info remains the same. Thanks, Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Fri, 17 May 2024 17:10:10 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: Fwd: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Reposting this. If there is an issue with HTML format and/or wireshark screen cap and I need to upload that separately somewhere else, please let me know. Thanks. -Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Thu, 09 May 2024 05:32:27 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: [screencap link at https://www.signalogic.com/images/wireshark_capture_timestamp_jump.png] In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff ----- End forwarded message ----- ----- End forwarded message -----

2 4

core children setting
by Alexis Fidalgo 24 May '24

24 May '24

Hello all, im not clear on the children setting, currently we’re listening in a single UDP port, i do understand that children are how many forks per listener ip:port, but i have this question, is this value tied on how many requests are received? In our case every requests takes an average of 80ms to be processed (there are a few external queries executed per request), so, example which value for children should i use if i have 200req/s, 300, 400? Right now im using 16 but i can see some cpu peaks in high traffic hours. thanks

2 2

"Contact does not exist in PCSCF usrloc" error in log
by Shantanu K Roy 23 May '24

23 May '24

Hello, We are trying to integrate Kamailio as the IMS server with our 5G core emulator by following the steps listed out at https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/. In our case, we have our own 5G core emulator instead of the Open5GS core. We are using sipp scripts to generate the SIP calls. The PCRF component of the 5G core is emulated by us and is capable of handling Diamater calls from Kamailio over the Rx interface. We have got the setup working to the point where the SIP Register calls are landing on the PCSCF which in turn is sending out AA Requests to PCRF and the PCRF is responding. However there is an internal error in the Kamailio log "4(3682133) ERROR: ims_qos [ims_qos_mod.c:1305]: w_rx_aar_register(): This contact does not exist in PCSCF usrloc - error in cfg file" Please help us in identifying if we are missing anything in the Kamailio set up. The packet capture and syslog excerpts from the VM where Kamailio is running are attached; here we are trying to perform SIP Registration for three SIP endpoints 001010123456791, 001010123456792 and 001010123456793. If there is any additional information required, please let us know. Regards, Shantanu

1 0

VoLTE/IMS third party registration issues
by ruben.iniguez＠vodafone.com 23 May '24

23 May '24

Checking Kamailio in a VoLTE environment we are finding issues to make it work. First issues arise when getting AS to extract the required info from third party registration, primarily IMPUs. Third party registration is sent to AS with IMPI in To header, preventing the AS from extracting the IMPU from the most obvious place, the To header, according to 3GPP 24.229: "Notification of Application Servers about registration status [...] the To header field, which shall contain a non-barred public user identity belonging to the service profile of the processed Filter Criteria. It may be either a public user identity as contained in the REGISTER request received from the UE or one of the implicitly registered public user identities in the service profile, as configured by the operator;" On the other hand, the third party registration can only embed in its body either the original request or the original response but not both (notice mutually exclusive "else if" statements in third_party_reg.c). Both are needed to extract Feature-Caps (from original request) and IMPUs (for instance from P-Associated-URI in original response). Therefore to extract that info the AS should subscribe to the reg info in S-CSCF, but such SUBSCRIBE is rejected with 403. From Registrar IMS SCSCF documenation it looks like this is not supported: "This function checks to see that a SUBSCRIBE request is authorised to subscribe to the particular identity. Only 3 entities can subscribe: The user agent to its own state The P-CSCF specified in the path header for that user Application Server (AS) not yet implemented" Therefore question to the community is how these limitations are overcome in existing VoLTE deployments with Kamailio? Any advice?

1 0

Kamailio IPSec module
by H Yavari 23 May '24

23 May '24

Hi all, I was reviewing the `ims_ipsec_pcscf` code and noticed that this module creates a pool of sockets using different ports (ipsec_max_connections). I'm unclear on the necessity of this approach. Can't we simply create one server listener and one client listener to handle all UE connections? If this is feasible, is there still a need to create the pool at startup? I'm curious if this is due to an architectural limitation or if the IPSec module can be modified to replace the current implementation with a more efficient one. Thank you for your insights. Best regards,Hossein

3 5

S-CSCF SiFC support
by ruben.iniguez＠vodafone.com 22 May '24

22 May '24

Hello In the context of an IMS deployment, does anybody know if Shared iFC mechanism is supported in Kamailio? I don't see configuration options for it, and just wanted to confirm if only regular iFCs (residing on HSS) are supported. Thank you

1 0

Assistance Required for VoNR Setup with Version 5.8.1
by venkateshcoff＠gmail.com 22 May '24

22 May '24

Hi, I am trying to set up VoNR with version 5.8.1, but I cannot find the configuration information and documentation anywhere. Has anyone tested a VoNR call? Thanks.

1 0

usrloc: keepalives not being sent to UDP endpoints for a very long time
by luke＠primevox.net 21 May '24

21 May '24

We've noticed an issue with UDP endpoints and the usrloc module not sending keepalive packets to them for a long time. Eventually, usrloc will begin doing it, but sometimes it's anywhere between 10 minutes and multiple hours before the endpoint starts receiving an OPTIONS packet. TCP and TLS endpoints do not have this issue - they being receiving OPTIONS after registration + ka_timeout elapsed. Has anyone else seen this? It can be very easily seen with a pcap on a device registering via UDP.

1 0

A question about Db_flatstore Module in Kamailio
by فرزانه سلطان زاده 20 May '24

20 May '24

Hi, Hope to find you well, I have a question about the 'db_flatstore' module in Kamailio. Here's my script for log rotation: for f in acc_cdrs_*.log; do mv -- "$f" "${f%.log}.temp" done docker exec -it pcscf kamcmd flatstore.k_rotate rm *.temp I want to know if the kamcmd command is blocking and the files can be removed after that or if I should wait for it to be done before removing the renamed files. Regards, Farzaneh

3 2

Kamailio multidomain IMS
by Valentin Christoph 20 May '24

20 May '24

Dear IMS enthusiasts :-) Has anyone experience with Kamailio IMS (CSCF) setups, where users with different domain names can be registered at one and the same S-CSCF? Is this feasible at all? I mean one IMS for users <user>@ims1.example.com, <user>@ims2.example.com, <user>@ims3.example.com,........ in one and the same S-CSCF, I-CSCF, P-CSCF. Thanks, Christoph

2 1

q_malloc.c:520 WARNINGs
by Joel Serrano 20 May '24

20 May '24

Hi, Does anyone know what this WARNING means or how to fix it? 2024-03-09T00:39:15.406674-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406823-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.406973-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407089-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407178-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407278-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) 2024-03-09T00:39:15.407375-05:00 csbc03 csbc[33024]: WARNING: {1 1 INVITE CALL-ID} <core> [core/mem/q_malloc.c:520]: qm_free(): WARNING: free(0) called from dialplan: dp_db.c: pcre2_free(206) It happens after running dp_translate(). These are my settings: debug=L_DBG debug=L_NOTICE memdbg=L_INFO memlog=L_INFO corelog=L_ERR mem_summary=12 log_stderror=no log_facility=LOG_LOCAL0 log_name="csbc" log_prefix="{$mt $hdr(CSeq) $ci} " mem_join=1 Any ideas? Thanks, Joel.

2 5

tm.t_uac_wait ACK does not have Max-Forwards header
by Juha Heinanen 20 May '24

20 May '24

Kamailio sends INVITE by tm.t_uac_wait RPC command: INVITE sip:test-0x55e7f1f78e70@192.168.187.215:5022 SIP/2.0 Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 To: <sip:test@test.tutpro.com> From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 CSeq: 1 INVITE Call-ID: 6649e78355ac2(a)test.tutpro.com Content-Length: 131 User-Agent: OpenSIPg SIP Proxy (5.9.0-0b01 (x86_64/linux)) Max-Forwards: 10 Contact: <sip:click2dial@192.168.187.215:5060> Content-Type: application/sdp .. sdp omitted UAS accepts the INVITE: SIP/2.0 200 Answering Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 To: <sip:test@test.tutpro.com>;tag=a3302e2a0898de3e From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 CSeq: 1 INVITE Call-ID: 6649e78355ac2(a)test.tutpro.com Server: baresip v3.12.0 (x86_64/Linux) Contact: <sip:test-0x55e7f1f78e70@192.168.187.215:5022> Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,UPDATE,REFER Supported: gruu,replaces,norefersub Content-Type: application/sdp Content-Length: 239 .. sdp omitted After receiving the response, Kamailio automatically sends ACK: ACK sip:test-0x55e7f1f78e70@192.168.187.215:5022 SIP/2.0 Via: SIP/2.0/UDP 192.168.187.215;branch=z9hG4bK6cf1.f1d087d6000000000000000000000000.0 From: <sip:click2dial@test.tutpro.com>;tag=2dd3d4389fe3d66c5cb97dc370ab7d7b-9c5a2a58 Call-ID: 6649e78355ac2(a)test.tutpro.com To: <sip:test@test.tutpro.com>;tag=a3302e2a0898de3e CSeq: 1 ACK User-Agent: OpenSIPg SIP Proxy (5.9.0-0b01 (x86_64/linux)) Content-Length: 0 The problem is that Max-Forwards header is missing from the ACK. Should I have something in config file to add it or is this a bug? -- Juha

1 3

Segfault on startup when using TLS in kamailio 5.7.4
by stefanr＠wave.com 17 May '24

17 May '24

Hi there, We are encountering consistent segfaults after rebooting our Kamailio instance with incoming traffic, specifically when using Kamailio 5.7.4. We think this issue did not occur with version 5.7.2, so it seems to have been introduced in either 5.7.3 or 5.7.4. Due to team bandwidth constraints and the potential impact on production traffic, we don't want to spend time on trying to reproduce the issue. So we have decided to downgrade to 5.6.4, which we confirmed to be stable. (Probably 5.7.2 would be too - but we didn't try). Unfortunately, our logging was only set to WARNING level, and we did not capture a core dump, so we cannot provide additional details beyond the following logs: This was with tcp_reuse_ports=yes: 2024-05-17T15:42:55.582475541Z Listening on 2024-05-17T15:42:55.582512370Z [redacted] 2024-05-17T15:42:55.582538161Z tls: 10.X.X.X:5061 advertise Y.Y.Y:5061 2024-05-17T15:42:55.582543750Z Aliases: 2024-05-17T15:42:55.582549081Z tls: [redacted]:5061 2024-05-17T15:42:55.582574890Z 2024-05-17T15:42:55.587876630Z 0(1) WARNING: tls [tls_init.c:978]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 18874368 and 9437184 bytes 2024-05-17T15:42:55.703927049Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 23 2024-05-17T15:42:55.703972029Z 0(1) ALERT: <core> [main.c:791]: handle_sigs(): child process 15 exited by a signal 11 2024-05-17T15:42:55.703978409Z 0(1) ALERT: <core> [main.c:795]: handle_sigs(): core was generated 2024-05-17T15:42:55.705049839Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 17 2024-05-17T15:42:55.705074209Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 21 2024-05-17T15:42:55.705081209Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 22 2024-05-17T15:42:55.705085879Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 20 2024-05-17T15:42:55.705090319Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 18 2024-05-17T15:42:55.705094649Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 19 2024-05-17T15:42:55.705098879Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 16 2024-05-17T15:42:55.705207399Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 15 2024-05-17T15:42:55.705459439Z 35(41) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 27 Without tcp_reuse_ports=yes, the segfault was always preceded by the following line if any existing TLS connections were stuck in TIME_WAIT: 2024-05-16T19:18:51.654447639Z 9(14) WARNING: {1 1 INVITE XXX(a)0.0.0.0} <core> [core/tcp_main.c:1301]: find_listening_sock_info(): binding to source address 10.X.X>X:5061 failed: Address already in use [98] 2024-05-16T19:18:51.746994728Z 0(1) ALERT: <core> [main.c:791]: handle_sigs(): child process 14 exited by a signal 11 When the server wasn't handling any traffic, the issue didn't occur even in 5.7.4. Does anyone have any insights or suggestions on how to address this issue? Kind regards Stefan

2 2

Assistance Needed to Resolve Duplicate Contact Headers in 200 OK Response
by satyaprakash ch 17 May '24

17 May '24

Hi We are receiving 2 to 3 contact headers in the 200 OK response of the registration request. It appears that the `user_location` is updating multiple contact headers for the same extension due to network disconnections. Please suggest a solution to avoid duplicates in the `user_location` module, or how to ensure that only the latest single contact header is included in the 200 OK response. Your suggestion would be appreciated. Thanks and regards, Satyaprakash.

2 1

Fwd: rtpengine timestamp jumps
by Jeff Brower 17 May '24

17 May '24

Reposting this. If there is an issue with HTML format and/or wireshark screen cap and I need to upload that separately somewhere else, please let me know. Thanks. -Jeff ----- Forwarded message from Jeff Brower <jbrower(a)signalogic.com> ----- Date: Thu, 09 May 2024 05:32:27 +0000 From: Jeff Brower <jbrower(a)signalogic.com> Subject: [SR-Users] rtpengine timestamp jumps To: sr-users(a)lists.kamailio.org Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff ----- End forwarded message -----

1 0

$dlg_var() not implemented in ims_dialog module
by Valentin Christoph 17 May '24

17 May '24

Dear all, I would probably need to store some dialog scoped information in an IMS environment, but the ims_dialog module does not implement the $dlg_var() pseudo variable. Do you know other, maybe better, possibilities to store dialog scoped information in an IMS (CSCF) environment? Thank, Christoph

3 2

making HTTP Async request in stateless configuration
by Sergio Charrua 17 May '24

17 May '24

Hi all! Just wondering what would be you opinion on the following. We are integrating Kamailio as a stateless redirect server, using SIP 300 Multiple Choices response. The script requires that prior to the redirection, a HTTP request must be made. We thought about using HTTP Async Client to make the requests (POST of JSON content), which is our prefered way of sending HTTP requests, and depending on the response from the REST API, reply back to UAC a SIP 300 Multiple Choices message and terminate the call flow on Kamailio. From what I know, HTTP Async Client requires that the TM module be used, which (it is my understanding) will imply that the whole Kamailio script to be stateful, which I think is useless as the SIP flow will be really simple: INVITE -----> <------- 100 Trying <------- 300 Multiple Choices ACK ------> There is a catch, though, the CPS is very high, above 1000 CPS.... Could anyone share their thoughts on this? Should we go stateful and use the HTTP Async module? Should we go stateless and use the HTTP client module? What would be the best approach to implementing this? Thanks guys! *Sérgio Charrua*

1 0

WebRTC-UDP voicemail and call transfer
by Social Boh 16 May '24

16 May '24

Hello list, I'm trying implementing a solution like: https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg The 2 problems I have a this moment are: Call from UDP phone to WebRTC phone, the WebRTC phone reject the call; the call go to voicemail but in the SDP still have WebRTC SDP During a SIP UDP - WebRTC Call tranfer the call using phone button... the transfer don't work. Any hint Regards -- --- I'm SoCIaL, MayBe

1 0

dynamic selection of authentication algorithm
by Juha Heinanen 14 May '24

14 May '24

How about this new function that sets the algorithm? -- Juha iff --git a/src/modules/auth/auth_mod.c b/src/modules/auth/auth_mod.c index 534ad9e20f..4e3a584d22 100644 --- a/src/modules/auth/auth_mod.c +++ b/src/modules/auth/auth_mod.c @@ -74,6 +74,10 @@ int w_consume_credentials(struct sip_msg *msg, char *s1, char *s2); * Check for credentials with given realm */ int w_has_credentials(struct sip_msg *msg, char *s1, char *s2); +/* + * Set authentication algorithm + */ +int w_auth_algorithm(struct sip_msg *msg, char *alg, char* s2); static int pv_proxy_authenticate( struct sip_msg *msg, char *realm, char *passwd, char *flags); @@ -170,6 +174,8 @@ static cmd_export_t cmds[] = { REQUEST_ROUTE}, {"pv_auth_check", (cmd_function)w_pv_auth_check, 4, fixup_pv_auth_check, 0, REQUEST_ROUTE}, + {"auth_algorithm", w_auth_algorithm, 1, fixup_spve_null, 0, + REQUEST_ROUTE}, {"bind_auth_s", (cmd_function)bind_auth_s, 0, 0, 0}, {0, 0, 0, 0, 0, 0} @@ -475,6 +481,35 @@ int w_has_credentials(sip_msg_t *msg, char *realm, char *s2) return -1; } return ki_has_credentials(msg, &srealm); + +} +/** + * + */ +int w_auth_algorithm(sip_msg_t *msg, char* alg, char* s2) +{ + if (fixup_get_svalue(msg, (gparam_t*)alg, &auth_algorithm) < 0) { + LM_ERR("failed to get algorithm value\n"); + return -1; + } + + if (strcmp(auth_algorithm.s, "MD5") == 0) { + hash_hex_len = HASHHEXLEN; + calc_HA1 = calc_HA1_md5; + calc_response = calc_response_md5; + } + else if (strcmp(auth_algorithm.s, "SHA-256") == 0) { + hash_hex_len = HASHHEXLEN_SHA256; + calc_HA1 = calc_HA1_sha256; + calc_response = calc_response_sha256; + } + else { + LM_ERR("Invalid algorithm provided." + " Possible values are \"\", \"MD5\" or \"SHA-256\"\n"); + return -1; + } + + return 1; } #ifdef USE_NC

3 6

302 Message Handling
by tfayomi＠untangledtechnology.com 14 May '24

14 May '24

Hi All, I'm having trouble understanding/finding resources on how to handle a 302 message I receive. I'm working on making an inbound sbc with kamailio with clearip integration. I'm successfully sending to kamailio, but am having trouble retrieving the response from clearip in kamailio. For some background, the intended call flow should be this: first it should send from caller to kamailio (caller > kamailio), then over tcp kamailio sends to clearip (kamailio > clearip) [This is what is working as expected] Then I should be able to signal to kamailio to continue the call to the intended destination. When clearip sends a code back (in this case 302), I want to be able to see that code and send the caller to the correct destination. The problem I keep running into is that I can't seem to see the 302 code in a reply, branch or failure root or checking the status. I'm just lost on what I am misunderstanding and would appreciate if anyone could point me in the right direction on how to approach retrieving the 302 message. I'm using kamailio version 5.6.3 and debian 12.1 Thanks in advance, Temi

2 1

Prefix match in htable
by Antonio Reale 13 May '24

13 May '24

Hello, I'm testing Kamailio 5.x as a redirecting server based on calling number. The calling numbers are stored on a DB and loaded in htable at runtime. When the INVITE or REGISTER arrive Kamailio checks if there is a match on $fU in htable (KEY_NAME) and if there is one, it replies with 302 adding the Contact HF with $fU@$sht(cli=>$fU)(with the domain part found in KEY_VALUE): append_to_reply("<Contact: sip:$fU@$sht(cli=>$fU)>\r\n"); If I use the full number as KEY_NAME it works as expected, but how can I match on a prefix instead of the whole $fU? For example, If I receive a call form 0655123456 I want to match it also if I have the prefix 0655 in htable. Do you have some advice to achieve this goal with htable or in a similar way? Thank you. Best regards. -- Antonio Reale

1 0

Kamailio and RTPengine and no voice on calls
by palany 10 May '24

10 May '24

I have an AWS ec2 instance with Kamailio and rtpengine configured. I am having a challenge of no voice on calls. I have sip client on my mobile phone and soft phone on my laptop and the two are on different networks behind nat. Signaling is working fine but there is no media and when I check on on the SDP the call logs are showing private ips instead of public ips : o=- 3 2 IN IP4 192.168.1.33 and c=IN IP4 172.31.36.50. I understand this is a NAT issue but I am failing to see where is the challenge. My rtpengine seems to be working fine . Below is my Kamailio.cfg file link : https://docs.google.com/document/d/1MKZJGlQjxYbdtw6QcCm9_EmvVPOkMs-YXjlq1ZM2 mmY/edit?usp=drive_link PCAP file for the call: https://docs.google.com/document/d/12gofLYhN0-lYIZORngIeuaYxLudLjiNdZgY504Ei XFs/edit?usp=drive_link rtpengine.conf https://docs.google.com/document/d/1QPxey49UzuN3Wd8jp8jc0mFhh31rr1alIO-gl4gP JEI/edit?usp=drive_link Please help.

2 3

base64 decoding issue
by Sergio Charrua 10 May '24

10 May '24

Hi all! I have been dealing with STIR/SHAKEN for a few weeks now, and while doing some tests I have found an issue which I can't find the reason for. The script has the following route logic: route[HANDLE_STIRSHAKEN] { xlog("L_INFO", "HANDLE_STIRSHAKEN - STIR/SHAKEN Logic"); if ($hdrc(Identity) == 0 ){ xlog("L_INFO", "HANDLE_STIRSHAKEN - No Identity Header -> Skipping this logic"); return; } #Verify Call Identity xlog("L_INFO", "HANDLE_STIRSHAKEN - Identity Header: $hdr(Identity)"); $var(header_1)= $(hdr(Identity){s.select,0,.}); $var(header_2)= $(hdr(Identity){s.select,1,.}); xlog("L_INFO", "HANDLE_STIRSHAKEN - Encoded Header 1: $var(header_1)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Encoded Header 2: $var(header_2)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 1: $(var(header_1){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 2: $(var(header_2){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Parsing!"); $var(hdr_1_json_payload) = $(var(header_1){s.decode.base64}); # Header 1 xlog("L_INFO", "HANDLE_STIRSHAKEN - Parsed!"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Header 1 payload = $var(hdr_1_json_payload) "); $var(hdr_2_json_payload) = $(var(header_2){s.decode.base64}); # Header 2 jansson_get_field($var(hdr_2_json_payload), "attest", "$var(attest_value)"); jansson_get_field($var(hdr_2_json_payload), "ppt", "$var(ppt_value)"); jansson_get_field($var(hdr_2_json_payload), "typ", "$var(typ_value)"); jansson_get_field($var(hdr_2_json_payload), "x5u", "$var(x5u_value)"); xlog("L_INFO", "HANDLE_STIRSHAKEN - $var(attest_value) $var(ppt_value) $var(typ_value) $var(x5u_value) "); } All it does is parse the Identity header received on the initial INVITE, decodes it (it is encoded in base64) and writes some JSON values on log. Nothing serious.... As the Identity header is divided into multiple parts, separated by a ".", I get the first 2 parts in 2 private variables, as per lines: $var(header_1)= $(hdr(Identity){s.select,0,.}); $var(header_2)= $(hdr(Identity){s.select,1,.}); Logs show correctly that both variables are populated: May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Encoded Header 1: eyJhbGciOiAiRVMyNTYiLCJwcHQiOiAic2hha2VuIiwidHlwIjogInBhc3Nwb3J0IiwieDV1IjogImh0dHBzOi8vY2VydGlmaWNh dGVzLmNsZWFyaXAuY29tL2IxNWQ3Y2M5LTBmMjYtNDZjMi04M2VhLWEzZTYzYTgyZWMzYS83Y2M0ZGI2OTVkMTNlZGFkYTRkMWY5ODYxYjliODBmZS5jcnQifQ May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 230 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Encoded Header 2: eyJhdHRlc3QiOiAiQSIsImRlc3QiOiB7InRuIjogWyIxNDA0NTI2NjA2MCJdfSwiaWF0IjogMTU0ODg1OTk4Miwib3JpZyI6IHsi dG4iOiAiMTgwMDEyMzQ1NjcifSwib3JpZ2lkIjogIjNhNDdjYTIzLWQ3YWItNDQ2Yi04MjFkLTMzZDVkZWVkYmVkNCJ9 However, when decoding both values: xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 1: $(var(header_1){s.decode.base64})"); xlog("L_INFO", "HANDLE_STIRSHAKEN - Decoded Header 2: $(var(header_2){s.decode.base64})"); only $var(header_2) gets decoded: May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 38 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: *HANDLE_STIRSHAKEN - Decoded Header 1:* May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 182 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: *HANDLE_STIRSHAKEN - Decoded Header 2: {"attest": "A","dest": {"tn": ["14045266060"]},"iat": 1548859982,"orig": {"tn": "18001234567"},"origid": "3a47ca23-d7ab-446b-821d-33d5deedbed4"}* As you can veri And to ease debugging, I have isolated the line that generates the issue: May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Parsing! May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 *May 10 09:09:42 kamailio1 kamailio[153201]: ERROR: {1 10 INVITE 1-1966540(a)10.20.0.1 <1-1966540(a)10.20.0.1>} <core> [core/lvalue.c:351]: lval_pvar_assign(): non existing right pvar* *May 10 09:09:42 kamailio1 kamailio[153201]: ERROR: {1 10 INVITE 1-1966540(a)10.20.0.1 <1-1966540(a)10.20.0.1>} <core> [core/lvalue.c:411]: lval_assign(): assignment failed at pos: (21,32-21,64)* May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/parser/msg_parser.c:324]: parse_headers(): flags=ffffffffffffffff May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 34 May 10 09:09:42 kamailio1 kamailio[153201]: DEBUG: {1 10 INVITE 1-1966540(a)10.20.0.1} <core> [core/pvapi.c:1418]: pv_printf_mode(): final buffer length 27 May 10 09:09:42 kamailio1 kamailio[153201]: INFO: {1 10 INVITE 1-1966540(a)10.20.0.1} <script>: HANDLE_STIRSHAKEN - Parsed! Even though the $var(header_1) contains a base64 encoded string, the decoder seems to fail. If I copy the value contained in $var(header_1) to an online decoder like https://www.base64decode.org/ the string gets decoded correctly. I have even modified the line : $var(header_1)= $(hdr(Identity){s.select,0,.}); to get the reverse index from the Identity string by doing $var(header_1)= $(hdr(Identity){s.select,-6,.}); It gets the same encoded string, same value (as expected), but the result is the same: Kamailio's s.decode.base64 fails with the same error. i'm not sure what assignment failed at pos: (21,32-21,64) could mean.... I don't think my code is wrong, as the same function is used on $var(header_2) value and it works, but maybe someone can correct me. Greatly appreciate your help. Cheers, *Sérgio Charrua*

1 1

Failover with dialog dmq while dialog state 2
by Björn Klasen 09 May '24

09 May '24

Hello @all as we want to enhance our VoIP-Plattform with DMQ functionalities for replication of Userlocation, Dialog and Hash Tables I made a lot of tests durings the last week. For testing I use two Debian 12 Server with Kamailio 5.7.4 (latest git) Both server have their own floating IP so in case one server fails the floating changes to the working server. This is done via keepalived On both servers Kamailio is active and use ip_free_bind=1 so both Kamailio instances can listen to all necessary IPs. So we have following network configuration: Server1: static IP: 192.168.253.201 floating IP: 192.168.253.200 Server2: static IP: 192.168.253.202 floating IP: 192.168.253.210 Now I test some calls while I trigger a failover, so Server1 takes over 192.168.253.10 Message Test Result CustomerA -> Proxy -> CustomerB Failover while Dialogstate 4 BYE is relayed correctly. Dialog is changing to state 5 CDR needs to be written manually -> no problem at all CustomerA -> Proxy -> CustomerB Failover while Dialogstate 2 200 OK is relayed correctly although no transaction exist an therefore no t_reply route is triggered. This is not a big deal because i can use default onreply_route so all necessary variables can be used via hash table. BUT Big Problem: Dialog state stays 2, so a following ACK is a bogus event and the dialog state is never changed to 4. Even worse: after 300 seconds the dialog is cleaned away by the timer, so I have no chance to do any kind of accounting. Test 2 is a real mess, because for me there is only one solution. I have to check after a failover on incoming replies to an INVITE that was handled on Server2 if these current dialog state is < 4 and cancel them. But I can't see how to do it? Are there other ways than using t_cancel? Or is it even some kind of a bug that a 200 OK does not trigger state 3 on dialog (although I have in mind that it has to be something to do with missing transaction data, that are not replicated because of missing replication capabilities of tm module)? If none of the above is possible I really asking myself of a real use case of dialog dmq replication. -- Björn Klasen, Teamleitung NGN VoIP-Backbone TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Gunnar Peter, Sven Schade, Carsten Tolkmit, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

2 1

How to set "set_dialog_profile" value for Leg-a and leg-b.
by satyaprakash ch 09 May '24

09 May '24

Hi, We are using kamailio and freeswitch, So when we make a call from A to B the call is going via "kamailio -> freeswitch" and "freeswitch -> kamailio". Please suggest me how i can set "set_dialog_profile" for both legs with unique value in kamailio config file. Thanks.

1 0

Is there any way to modify the Refer-To header username?
by Shah Hussain Khattak 09 May '24

09 May '24

Hello Experts, I wanted to check, is there any way to modify the username part of the Refer-To header in Kamailio? For example, I am getting the following header: REFER-TO: <sip:sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> the username part is missing. Can I add a username inside it? For example, change it to REFER-TO: <sip:123@sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> REFER-TO: <sip:test-kam@sip.abc.domain.com:5061;transport=tls; ..... ; .... ; ...> I could not find any solution in the mailing list, the only way was to remove the header and then re-add it. Thank you! Regards, Shah Hussain

2 2

rtpengine timestamp jumps
by Jeff Brower 09 May '24

09 May '24

Hi rtpengine experts, We have some customers processing long multi-party call pcaps using mediaMin who are reporting large amounts of packets with timestamp jumps but no packet loss (for instance 10% of packets over a 1 hr 45 min call). For example, in the Wireshark excerpt shown below, packets 6 and 8 sent by rtpengine show a timestamp increment of 640, but sequence number increment of 1: In the mediaMin output packet log we typically see sections similar to: : : Seq num 98584 timestamp = 3902252372, rtp pyld len = 33 media-R Seq num 98585 timestamp = 3902252692, rtp pyld len = 33 media Seq num 98586 timestamp = 3902253012, rtp pyld len = 33 media-R Seq num 98587 timestamp = 3902253332, rtp pyld len = 33 media Seq num 98588 timestamp = 3902253652, rtp pyld len = 33 media-R Seq num 98589 timestamp = 3902253972, rtp pyld len = 33 media Seq num 98590 timestamp = 3902254292, rtp pyld len = 33 media Seq num 98591 timestamp = 3902254612, rtp pyld len = 33 media-R Seq num 98592 timestamp = 3902254932, rtp pyld len = 33 media Seq num 98593 timestamp = 3902255252, rtp pyld len = 33 media Seq num 98594 timestamp = 3902255572, rtp pyld len = 33 media-R Seq num 98595 timestamp = 3902255892, rtp pyld len = 33 media Seq num 98596 timestamp = 3902256212, rtp pyld len = 33 media Seq num 98597 timestamp = 3902256532, rtp pyld len = 33 media Seq num 98598 timestamp = 3902256852, rtp pyld len = 33 media-R Seq num 98599 timestamp = 3902257172, rtp pyld len = 33 media Seq num 98600 timestamp = 3902257492, rtp pyld len = 33 media-R Seq num 98601 timestamp = 3902257812, rtp pyld len = 33 media Seq num 98602 timestamp = 3902258132, rtp pyld len = 33 media Seq num 98603 timestamp = 3902258452, rtp pyld len = 33 media Seq num 98604 timestamp = 3902258772, rtp pyld len = 33 media-R Seq num 98605 timestamp = 3902259092, rtp pyld len = 33 media : : where media-R packets are timestamp gap repairs (i.e. frame loss concealment). The behavior tends to be bursty, but once it gets going it goes for a while and seems relatively consistent. Is this expected behavior for rtpengine ? If so is rptengine in turn dealing with some type of "slow packet rate" issue from a remote sender ? Thanks, Jeff

1 0

Kamailio v5.7.5 Released
by Daniel-Constantin Mierla 08 May '24

08 May '24

Hello, Kamailio SIP Server v5.7.5 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.5. For more details about version 5.7.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/05/kamailio-v5-7-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.7.x is currently the previous stable release series, the latest is 5.8.x and v5.8.1 was released on April 3, 2024. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v5.7.5
by Daniel-Constantin Mierla 08 May '24

08 May '24

Hello, I am considering to release Kamailio v5.7.5 (out of branch 5.7) on Wednesday, May 8, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

sql_query fails from event_route
by Juha Heinanen 07 May '24

07 May '24

I noticed that sql_query does not (anymore) work when called from event route: event_route [tcp:closed] { # Handle TCP connection close # Delete possible leftover registration if ($conid) sql_query("sip_proxy_cache", "DELETE FROM location WHERE server_id=0 AND connection_id=$conid", "result"); } The query produces errors: 2024-05-07T11:30:14.785915+03:00 siika /usr/bin/sip-proxy[2848078]: CRITICAL: sqlops [sqlops.c:213]: sql_check_connection(): no database handle with reconnect disabled 2024-05-07T11:30:14.785978+03:00 siika /usr/bin/sip-proxy[2848078]: ERROR: sqlops [sqlops.c:232]: sql_query(): invalid connection to database 2024-05-07T11:30:14.786013+03:00 siika /usr/bin/sip-proxy[2848078]: ERROR: sqlops [sqlops.c:237]: sql_query(): with query [DELETE FROM location WHERE server_id=0 AND connection_id=7] I added the last debug message in order to find out where the error comes from. The same query works fine when executed from regular route. Any ideas why the query fails from event_route? -- Juha

2 3

How to use environment variable
by Pavan Kumar 07 May '24

07 May '24

Hi everyone, I am trying to assign environment variable as follows listen=udp:0.0.0.0:5060 advertise $env(MY_IP):5060 Looks like using the environment variable as above is an invalid configuration. Is there a way to use IP from env var to advertise. Even better, is there a way to use result in a stun query as an advertised address? Thank you, Pavan Kumar

2 3

Retransmission behaviour after ACK
by Sergio Charrua 06 May '24

06 May '24

Hi all! I'm testing some redirect scenarios with Kamailio 5.7.4 and using Astterisk 13 and SIPp for unit testing. The kamailio's script is simple and uses TM, HTTP_ASYNC_MODULE, JSONRPC, JANSSON, RTJSON and XLog modules. Request route is simple: request_route { route(HANDLE_OPTIONS); route(HANDLE_RETRANS); route(HANDLE_CANCEL); route(REQINIT); route(HANDLE_INVITE); } where HANDLE_RETRANS is: route[HANDLE_RETRANS]{ # handle retransmissions xlog("L_INFO","HANDLE_RETRANS - message type is $mt for method $rm"); if (!is_method("ACK")) { if(t_precheck_trans()) { xlog("L_INFO","HANDLE_RETRANS - Precheck Trans"); t_check_trans(); xlog("L_INFO", "HANDLE_RETRANS - Exiting after Retransmission check - method $rm"); exit; } xlog("L_INFO","HANDLE_RETRANS - Check Trans"); t_check_trans(); } } and HANDLE INVITE is: route[HANDLE_INVITE]{ if (is_method("INVITE")) { xlog("L_INFO","MAIN - calling TO_CARRIER"); setflag(FLT_ACC); route(TO_CARRIER); exit; } } The HTTP requests call a REST API returning a JSON object, that is iterated and produces a CONTACT header containing multiple choices using the append_branch() command. The RELAY route is : route[RELAY] { # Sends a 300 Multiple Choices back to the proxy that requested the routing lookup xlog("L_INFO","REPLY_302 - send reply"); send_reply("300", "Multiple Choices"); xlog("L_INFO","REPLY_302 - exit"); exit; } Which returns indeed a Contact header with multiple contact choices. When using Asterisk to originate a call to Kamailio, the SIP call flow is as follows and terminates (SNGrep and TCPDump do not detect anything else): SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> However when using SIPp I get some retransmissions after the final ACK. SRC DST INVITE (SDP) ──────────────────────────> 100 trying -- your call is <────────────────────────── 300 Multiple Choices <────────────────────────── ACK ──────────────────────────> 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── 300 Multiple Choices <<<──────────────────────── SIPp scenario is : <?xml version="1.0" encoding="ISO-8859-1" ?> <!DOCTYPE scenario SYSTEM "sipp.dtd"> <scenario name="Basic Sipstone UAC">   <send retrans="500" start_rtd="true" rrs="true"> <![CDATA[ INVITE sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]> Contact: sip:sipp@[local_ip]:[local_port];transport=[transport]> Call-ID: [call_id] CSeq: 102 INVITE User-Agent: SIPp Supported: replaces, timer Content-Type: application/sdp Subject: Performance Test Session-Expires: 3600;refresher=uas Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=SIPp c=IN IP[media_ip_type] [media_ip] t=0 0 m=audio [media_port] RTP/AVP 0 a=rtpmap:0 PCMU/8000 ]]> </send> <recv response="100" rss="true"> </recv> <recv response="300" rtd="true" rss="true"> <action> <ereg regexp="\<([a-zA-Z0-9@.=;]+)[^>]+" search_in="hdr" header="Contact: " assign_to="1"/> <log message="INFO: 300 Contact Header: [$1]" /> <exec command="echo [$1] >> ./300_ContactHeader.txt" /> </action> <action> <ereg regexp="sip:.*@([0-9A-Za-z\.]+):([0-9]+);transport=([A-Z]+)" search_in="hdr" header="Contact:" check_it="true" assign_to="dummy,host,port,transport" /> </action> </recv>   <send rrs="true"> <![CDATA[ ACK sip:[call_number]@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Max-Forwards: 70 From: <sip:sipp@[local_ip]:[local_port]>;tag=[call_number] To: <sip:[call_number]@[remote_ip]:[remote_port]>[peer_tag_param] Contact: <sip:sipp@[local_ip]:[local_port]> Call-ID: [call_id] CSeq: 102 ACK User-Agent: SIPp Content-Length: 0 ]]> </send>   </scenario> The question is : is there anything wrong with my kamailio script? or with SIPp scenario? why does kamailio have different behaviours between Asterisk and SIPp? Any option to disable retransmissions on Kamailio ? (using UDP) Thanks in advance, *Sérgio Charrua*

2 3

topos with mask_callid
by Marrold 05 May '24

05 May '24

Hi all, I am using Kamailio 5.8.1 on a Debian 12 machine with topos and topoh hiding the topology. I have observed with modparam("topos", "mask_callid", 1), in-dialog requests are relayed with the incorrect caller-id. In the scenario, A calls B. The call is set up and the call-id for the B-Leg is successfully masked. Then, when A sends a BYE, the BYE is incorrectly relayed to B with the call-id of the A leg, and B replies with a 481 If I disable masking the call-id, or disable topos/topoh entirely, everything looks and works as it should. I have tested with 5.7.4 and the behaviour is the same. Does anyone have any ideas for where I should start looking? Thanks Matthew

1 0

Re: IMS SCSCF redundancy
by Oleg Belousov 03 May '24

03 May '24

Hi. I have a question re: IMS SCSCF redundancy.:There are several SCSCF instances, registration happens via SCSCF1, it goes down after that. UE make MO call.. PCSCF resolves the common SCSCF fqdn and sends an invite to available SCSCF2 which does not have UE profile as well as AS address (which is received during registration SAR/SAA in iFC). There are 2 options here, either to release the call (as per 3gpp spec upon specific release code, eg 408/503 UE should perform re-registration - not happen in our case unfortunately), or to send request for iFC (suspend the call, fire SAR, receive SAA, check available services, etc) and upon success continue the call (to AS first)/ Lke IMS Registrar SCSCF save function, but for the invite. To continue the call without interaction with AS is not a good idea.. Any advice or workaround on how to achieve desired SCSCF redundancy ? -- obelousov.tel

1 0

http_async_client and sl_send_reply()
by Masud Muborakshohi 02 May '24

02 May '24

Hello! I have a problem when sending a response sl_send_reply("415", "The other party does not support video calling!"); in route[HTTP_REPLY]. When I make a request using http_async_query() to send a push notification to a mobile device, I need to process the response from the push server and respond with a SIP message to the client. Before sending the request I do $http_req(suspend) = 0; What am I doing wrong? Why are $fU and $ru equal to null ? #processing HTTP responses route[HTTP_REPLY] { if ($http_ok && $http_rs == 200) { xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n"); } else if($http_rs == 415 || $http_rs == "415") { xlog("L_ERR", "route[HTTP_REPLY]: caller: $fU phone: $ru error: $http_err)\n"); send_reply("415", "The other party does not support video calling!"); } else { xlog("L_ALERT", "route[HTTP_REPLY]: error $http_err)\n"); } #exit; }

3 5

No ACK from Kamailio
by Arseniy Moskvich 02 May '24

02 May '24

Hi everyone. We are new to Kamailio and using Kamailio 5.8 with FreeSwitch. We have the issue with the incoming call from our provider. They are dropping after 30 seconds. The client doesn't receive ACK from Kamailio. But FreeSwitch sends ACK to Kamailio. The outbound call is working fine. At the moment flow is our provider -> FreeSwitch -> Kamailio -> our web client. Could someone give us some directions please? Please find attached Kemi.py. Thank you so much in advance

2 2

Kamailio works but voice is not present during the calls!
by christian.marinelli＠hotmail.it 02 May '24

02 May '24

Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it. Can anyone help me to understand why? I can copy and paste here all the config files if you need it. Thank you so much in advance Christian

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