- sr-users - kamailio.org

Strange memory stats
by Barry Flanagan 30 Jan '25

30 Jan '25

Hi, I have a 5.8.4 runing on Bookworm that after a while reports very strange memory stats like this: shmem.fragments: '729' shmem.free_size: '18446744009887078560' shmem.max_used_size: '67043934736' shmem.real_used_size: '67043698528' shmem.total_size: '3221225472' shmem.used_size: '67034971568' The settings are: SHM_MEMORY=3072 PKG_MEMORY=256 The shmem.free_size and max_used/max_size seem quite wrong. The output of "mod.stats all shm" shows the below: - 'Module: jsonrpcs' - Total: 0 - 'Module: core' - Total: 7891296 - 'Module: rr' - Total: 0 - 'Module: dialog' - Total: 146144 - 'Module: permissions' - Total: 130672 - 'Module: auth' - Total: 0 - 'Module: topoh' - Total: 0 - 'Module: sl' - Total: 14048 - 'Module: siptrace' - Total: 79184 - 'Module: sqlops' - Total: 0 - 'Module: acc' - Total: 0 - 'Module: xlog' - Total: 0 - 'Module: dmq' - Total: 1520 - 'Module: avpops' - Total: 0 - 'Module: rtpengine' - Total: 69328 - 'Module: db_mysql' - Total: 0 - 'Module: geoip2' - Total: 32 - 'Module: pv' - Total: 0 - 'Module: registrar' - Total: 0 - 'Module: textops' - Total: 0 - 'Module: htable' - Total: 6375584 - 'Module: tsilo' - Total: 74048 - 'Module: usrloc' - Total: 554240 - 'Module: tm' - Total: 6812224 - 'Module: websocket' - Total: 2551536 - 'Module: pike' - Total: 6880 - 'Module: tls' - Total: 5462960 - 'Module: tmx' - Total: 21968 - 'Module: kex' - Total: 3536 - 'Module: cfgutils' - Total: 128 - 'Module: sipdump' - Total: 32 - 'Module: domain' - Total: 5920 - 'Module: dispatcher' - Total: 20144 Thanks! -Barry

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Kamailio v6.0.0 Released - new major version is out
by Daniel-Constantin Mierla 29 Jan '25

29 Jan '25

Kamailio v6.0.0 is out – it comes with 5 new modules and important updates to the core components and existing modules. The increment of the first version number reflects several important updates, among them: multi-threaded option for UDP receiving, option for specifying a range of ports to listen to, support for expressions with variables for most of the module functions parameters, internal libraries are directly inked to the core, support to use cmake to compile and install Kamailio. You can read a bit more detailed release notes at: * https://www.kamailio.org/w/kamailio-v6.0-0-release-notes/ Many thanks to all developers and community members that made possible this release. v6.0.0 brings more flexibility and optimizations across many existing components and modules, new parameters, functions, variables and transformations. Enjoy Kamailio v6.0.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

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Releasing v5.8.5
by Daniel-Constantin Mierla 29 Jan '25

29 Jan '25

Hello, I am considering to release Kamailio v5.8.5 (out of branch 5.8) on Thursday, Jan 23, 2025. 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

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MS Teams BYE routing
by Barry Flanagan 28 Jan '25

28 Jan '25

Hi all, I have a Teams gw working apart from one issue., In a call originated in Teams,. if the called party hangs up, the BYE is sent to api-du-b-euwe.pstnhub.microsoft.com:443 which does not work. This is taken from the Contact header of the ACK I presume. Here is the ACK to the 200 OK: 28/01/2025 17:08:26.560 +00:00: 52.114.75.24:53728 -> 100.100.198.241:5061 ACK sip:9000@teams.domain.com:5061;transport=tls;tps=atpsh-67990f01-8ed8-1 SIP/2.0 FROM: "Teams User" <sip:+35314375362@sip.pstnhub.microsoft.com:5061;user=phone>;tag=c29cb7a79ddf40f9ba909f3f1034a770 TO: <sip:9000@teams.domain.com:5061>;user=phone;tag=as6e287620 CSEQ: 1 ACK CALL-ID: e49f7df631ee576286ecf159e81952fe MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK46d28f33 CONTACT: <sip:api-du-b-euwe.pstnhub.microsoft.com:443;x-i=0781e8a1-91e6-4f6a-bc4a-73d6890827d4;x-c=e49f7df631ee576286ecf159e81952fe/d/8/703117d8337e4975a8216a54f59035c2> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2025.1.20.11 i.EUWE.7 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY ..and here is the BYE. 28/01/2025 17:08:36.119 +00:00: 100.100.198.241:5060 -> 52.123.134.13:443 BYE sip:api-du-b-euwe.pstnhub.microsoft.com:443;x-i=0781e8a1-91e6-4f6a-bc4a-73d6890827d4;x-c=e49f7df631ee576286ecf159e81952fe/d/8/703117d8337e4975a8216a54f59035c2 SIP/2.0 Record-Route: <sip:teams.domain.com:5061;transport=tls;ftag=as6e287620;lr;r2=on> Record-Route: <sip:100.100.198.241:5060;ftag=as6e287620;lr;r2=on> Via: SIP/2.0/UDP 100.100.198.241;branch=z9hG4bKc89c.0c1cf04a771c30293432e5f1f59331c8.0 Via: SIP/2.0/UDP 102.102.139.182:5060;rport=5060;branch=z9hG4bK3c42c163 Max-Forwards: 69 From: <sip:9000@teams.domain.com:5061>;user=phone;tag=as6e287620 To: "Teams User"<sip:+35314375362@sip.pstnhub.microsoft.com:5061;user=phone>;tag=c29cb7a79ddf40f9ba909f3f1034a770 Call-ID: e49f7df631ee576286ecf159e81952fe CSeq: 102 BYE Content-Length: 0 How can I get around this? Using a pretty standard WITHINDLG route. I am using topos but not manipulating the Contact header. Thanks! -Barry

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RTPengine module - pause/resume recording
by Who AmI 28 Jan '25

28 Jan '25

Hi list, I have been looking at the recording daemon in RTPengine with Kamailio controlling this via rtpengine_manage() flags: "record-call=on" in the initial Invite, or 200OK based on direction. I tested and this works brilliantly for the full calls, but when looking at pause/resume the module documentation is bare and I am a bit confused. It looks like the ngcp docs and Opensips module docs (for RTPengine) describe the pause/resume as just another start/stop flag. Use case is simple, all calls are recorded, but we want the option to pause, then resume once any private or personal information exchange has completed. My question is also simple - Is this the recommended way to pause and then resume recordings or have I missed something? The Kamailio documentation doesn't explicitly say this ( https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html#rtpengi…) so just looking for advice if this is the best route to go. Thanks, John.

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Re: [sr-dev] Re: Branching 6.0 series
by Daniel-Constantin Mierla 28 Jan '25

28 Jan '25

Hello, On 20.01.25 08:25, Daniel-Constantin Mierla via sr-dev wrote: > Hello, > > I think we should branch it on Wednesday, Jan 22, 2025, no matter on > what stage we are with cmake support. The old-Makefiles should be kept > anyhow (as they are or in a special folder with an easy way to recover > them), because there are many bits and pieces that can be discovered > later when more users. > > Because there were no many bugs reported to the C code specific to 6.0, > we can aim releasing on January 29, 2025, with old-Makefile still to be > used for tasks that are not yet covered by cmake. shall we still aim for releasing v6.0.0 on this upcoming Wednesday, January 29, 2025? We the old-Makefiles kept, it is doable, the wiki pages needs to be created for this release series, which I can do tomorrow, as I waited to see if how it should be the recommended method for compilation/installation. If the feeling is that we should wait, then probably the target date has to be shifted somewhere in the week starting on February 10, 2025, because Fosdem follows this weekend and then some countries have a week holiday break. If no other opinions and change of decision, the release will be done on Wednesday, with content using the old-Makefiles (to reuse existing tutorials) and reference to cmake variant when it is the case. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

2 1

defining configuration variables
by Sergio Charrua 27 Jan '25

27 Jan '25

Hello, just wondering if there is any way to define 2 constants as : # sets quarantine period the URL is set, in secs, after number of HTTP_RETRIES retries #!define SOME_VAR_1 60 # sets the configuration values for HTable module parameter #!define SOME_VAR_2 "some_configration_value" + SOME_VAR_1 I couldn't find anything in docs, and made several different ways without success. Any suggestions? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

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UDP worker lock investigation
by Володимир Іванець 26 Jan '25

26 Jan '25

Hello everyone! I was wondering if it is possible to get information about a packet that is being possessed by the worker without increasing log level? I have an installation with Kamailio that is configured with two interfaces. Once in a while Kamailio stops accepting requests on one of the interfaces. It works fine for requests from other one. I think that worker processes might be locked while processing some requests. Enabling debug log is not an option because the traffic is quite high and the log file will not just be enormous, some lines will be missing. It looks like syslog will skip some of the log entries. So is it possible to check what packet worker process is currently processing? Or do I need to add some log entries with lower log level to trace requests manually? Thank you!

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How to access registration contact user agent string in branch route after location lookup?
by Benoit Panizzon 24 Jan '25

24 Jan '25

Hi List I have some CPE which require special handling like removing 100rel and update from the supported methods because they misbehave. Previously, I was using reg_fetch_contacts and then appending the branches manually and stacking the UA in an AVP. With the branch index this allowed me to get the AVP containing the UA to which the call was branched in the branch route and perform filtering of methods based on UA, but it was complicated. Now I use the intended function lookup_to_dset() but now I miss how to access the register contact UA from within a branch route. Is there an intended way to access that information? 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 ______________________________________________________

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Use local_identity or custom data to set callerID
by matu lopi 23 Jan '25

23 Jan '25

Hi team, Is it possible to use either local_identity or custom data properties to define the CallerID during the RTCSession lifecycle? Thank you.

2 1

Issue with ACK on INVITE
by Barry Flanagan 23 Jan '25

23 Jan '25

Hi folks, Trying to hook up a new endpoint but am having an issue. Kamailio is in front of an Asterisk box. They send an INVITE, we send 100, then 200 OK. However, when they send their ACK, the RURI is not set to the Contact of the 200, instead it is<number>@<proxy ip>. This causes the ACK to get routed to the proxy itself and the call fails. 82.160.190.100:5060 = Kamailio IP Port 198.201.240.242:5080 = Asterisk IP/Port 182.33.174.10:5060 = Provider endpoint The far endpoint say they cannot fix the RURI - should I be able to handle this ACK below? My understanding is the ACK's RURI should be the Contact of the 200 OK. 200 OK sent from us to the Provider (Contact shows correct URI) ========================================================= 2025/01/15 17:14:49.470634 82.160.190.100:5060 -> 182.33.174.10:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 182.33.174.10;branch=z9hG4bKefc7.ad07f2d3.0 Via: SIP/2.0/UDP 182.33.174.39:5060;received=182.33.174.39;branch=z9hG4bK5bbf9591;rport=5060 Record-Route: <sip:1800715080@82.160.190.100:5061;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes> Record-Route: <sip:1800715080@82.160.190.100;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes> Record-Route: <sip:182.33.174.10;lr;ftag=as0b42eef3> From: "0737965510" <sip:0737966610@182.33.174.39>;tag=as0b42eef3 To: <sip:1800715080@82.160.190.100>;tag=as4133584e Call-ID: 480dc17b3f88516f364778dc4b2528da@182.33.174.39:5060 CSeq: 102 INVITE Server: Asterisk PBX 18.9.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces *Contact: <sip:1800715080@198.201.240.242:5080>* Content-Type: application/sdp Content-Length: 314 , ACK Reply from Provider to Us (RURI points to Kamailio ip:port now =========================================================== 2025/01/15 17:14:49.478119 182.33.174.10:5060 -> 82.160.190.100:5060 *ACK sip:1800715080@82.160.190.100:5060 *SIP/2.0 Via: SIP/2.0/UDP 182.33.174.10;branch=z9hG4bKefc7.ad07f2d3.2 Via: SIP/2.0/UDP 182.33.174.39:5060;received=182.33.174.39;branch=z9hG4bK55656469;rport=5060 Route: <sip:1800715080@82.160.190.100;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes>,sip:1800715080@82.160.190.100:5061;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes> Max-Forwards: 69 From: "0737965510" <sip:0737966610@182.33.174.39>;tag=as0b42eef3 To: <sip:1800715080@82.160.190.100>;tag=as4133584e Contact: <sip:0737965510@182.33.174.39:5060> Call-ID: 480dc17b3f88516f364778dc4b2528da@182.33.174.39:5060 CSeq: 102 ACK Thanks -Barry

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Kamailio v5.8.5 Released
by Daniel-Constantin Mierla 23 Jan '25

23 Jan '25

Hello, Kamailio SIP Server v5.8.5 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.4. 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 v5.8.5. For more details about version 5.8.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/01/kamailio-v5-8-5-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 World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

Getting headers from xhttp/websocket
by Barry Flanagan 23 Jan '25

23 Jan '25

Hi, I need to be able to pull in a header provided by a webrtc endpoint so I can use it in my request_route. I understand that I will need to use shm/htable to make it visible but I am unable to get the xhttp header using $header(x-p-push), or any header by name. In my event_route[xhttp:request] I have the following: if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET") { .... xlog("L_INFO", "BF x-p-push header is $hdr(x-p-push) Contact is $hdr(Contact)\n"); } The log shows both x-p-push and Contact header as null. sipdump shows the headers present in the request. Am I missing something? Running Kamailio 5.8.4 Thanks -Barry

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Development is open for future release of v6.1.x series
by Daniel-Constantin Mierla 22 Jan '25

22 Jan '25

Hello, the branch 6.0 was created, therefore the master branch is open for adding new features, to be part of future release series v6.1.x. Any bug fix committed to master that applies to 6.0.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 6.0 or 5.8. Expect that v6.0.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v6.1.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime during the last part of 2025 or the beginning of 2026. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

1 0

Branch 6.0 has been created
by Daniel-Constantin Mierla 22 Jan '25

22 Jan '25

Hello, the branch 6.0 has been created, to be used for releasing v6.0.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-6.0 cd kamailio-6.0 git checkout -b 6.0 origin/6.0 Pushing commits in this branch: git push origin 6.0:6.0 Note that 6.0 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 6.0 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 6.0 will be out, respectively Kamailio v6.0.0. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Not able to get the $TV(un) inside xhttp module
by kaseenadh111＠gmail.com 22 Jan '25

22 Jan '25

Hi, I am running Kamailio 5.7.* RHEL I need to log the latency of each XHTTP request (milli or microseconds precision) so I tried using $TV(un) which always gives a 0 value. Can we access $TV(un) under XHTTP routes? event_route[xhttp:request] { $var(tv_http_req_start_time) = $TV(un); xlog("L_INFO", "$$var(tv_http_req_start_time)= $var(tv_http_req_start_time) and actual $$TV(un)= $TV(un)"); .... } Output: Jan 22 16:18:20 VM-123************ kamailio[346535]: INFO: <script>: $var(tv_http_req_start_time)= 0 and actual $TV(un)= 0 $TS - I have tried this, it gives the Unix in seconds precision but I want milliseconds precision. Thanks Kasee

1 0

Re: [sr-dev] Re: Branching 6.0 series
by Daniel-Constantin Mierla 22 Jan '25

22 Jan '25

Hello, later today, in a few hours, I am going to create the branch 6.0. More details will be provided once it has been created. Cheers, Daniel On 20.01.25 08:25, Daniel-Constantin Mierla via sr-dev wrote: > Hello, > > I think we should branch it on Wednesday, Jan 22, 2025, no matter on > what stage we are with cmake support. The old-Makefiles should be kept > anyhow (as they are or in a special folder with an easy way to recover > them), because there are many bits and pieces that can be discovered > later when more users. > > Because there were no many bugs reported to the C code specific to 6.0, > we can aim releasing on January 29, 2025, with old-Makefile still to be > used for tasks that are not yet covered by cmake. > > Cheers, > Daniel > > On 15.01.25 11:16, Henning Westerholt via sr-dev wrote: >> Hello, >> >> yes, Xenofon is also waiting for a final reply from Sergey for the RPM packaging if now all is fine (was discussed as part of PR #4085). >> >> Cheers, >> >> Henning >> >>> -----Original Message----- >>> From: Victor Seva via sr-dev <sr-dev(a)lists.kamailio.org> >>> Sent: Mittwoch, 15. Januar 2025 10:59 >>> To: sr-dev(a)lists.kamailio.org >>> Cc: Victor Seva <linuxmaniac(a)torreviejawireless.org> >>> Subject: [sr-dev] Re: Branching 6.0 series >>> >>> Hi, >>> >>> please postpone. I didn't have time finalize the test of cmake builds on debian >>> packaging yet. >>> >>> On 15/1/25 7:18, Daniel-Constantin Mierla via sr-dev wrote: >>>> Hello, >>>> >>>> creating git branch 6.0 was planned for today and I wonder if it shall >>>> be done latter during the evening or it would be preferred to be >>>> postponed for two days or so, till end of the week/Friday, as I >>>> noticed yesterday some new work being done to cmake files for FreeBSD >>>> and maybe new commits are still coming. >>>> >>>> Of course, there is also the option to create the branch and then the >>>> new commits backported, but maybe postponing makes everything more >>>> convenient. >>>> >>>> In short, if anyone feels that postponing a bit creating the branch >>>> 6.0 simplifies their current work, let us know. If nobody ask for >>>> postponing, the branch will be created later towards the end of today. >>>> >>>> Cheers, >>>> Daniel >>> -- >>> ----------------------------------------------------------------- >>> | ⢀⣴⠾⠻⢶⣦⠀ Victor Seva | >>> | ⣾⠁⢠⠒⠀⣿⡁ linuxmaniac(a)torreviejawireless.org | >>> | ⢿⡄⠘⠷⠚⠋⠀PGP: 8F19 CADC D42A 42D4 5563 730C 51A0 9B18 CF5A >>> 5068 | >>> | ⠈⠳⣄⠀⠀⠀ Debian Developer | >>> ----------------------------------------------------------------- >> _______________________________________________ >> Kamailio - Development Mailing List -- sr-dev(a)lists.kamailio.org >> To unsubscribe send an email to sr-dev-leave(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to the sender! > > -- > Daniel-Constantin Mierla (@ asipto.com) > twitter.com/miconda -- linkedin.com/in/miconda > Kamailio Consultancy, Training and Development Services -- asipto.com > > _______________________________________________ > Kamailio - Development Mailing List -- sr-dev(a)lists.kamailio.org > To unsubscribe send an email to sr-dev-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender! -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

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Multi-line Via: rport added incorrectly
by James Browne 22 Jan '25

22 Jan '25

Hi I see that when kamailio adds rport to the Via header field of a request that has two Via values on the same line (comma-separated, of course), it adds the rport (and received) to the wrong value. I have this test kamailio.cfg for demonstration. children=1 loadmodule "sl" loadmodule "textops" loadmodule "nathelper" request_route { force_rport(); sl_send_reply(200, "OK"); } I send this OPTIONS (truncated) and get this response (truncated). The rport should be on the first Via value, not the second. OPTIONS sip:server SIP/2.0 Via: SIP/2.0/UDP 3.9.4.2:5060;branch=z9hG4bK503983089,SIP/2.0/UDP 1.3.4.1:5060;branch=z9hG4bK53805983059 From: <sip:client>;tag=LeonhardEuler To: <sip:server> SIP/2.0 200 OK Via: SIP/2.0/UDP 3.9.4.2:5060;branch=z9hG4bK503983089,SIP/2.0/UDP 1.3.4.1:5060;branch=z9hG4bK53805983059;rport=53805;received=127.0.0.1 From: <sip:client>;tag=LeonhardEuler To: <sip:server>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.49678e65 Server: kamailio (6.0.0-pre0 (x86_64/linux)) I've tested this with the latest commit (c994fb8) from kamailio. I can't think how this can be anything other than a bug. Should I log a bug report for this? James

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Error when sending NOTIFYs to disconnected websocket clients
by Flavio Grossi 20 Jan '25

20 Jan '25

Hi, i have some SIP websocket clients (built using the SIP.js library) SUBSCRIBEd to other clients via kamailio v5.8.2. If one of the watchers suddenly disappears (e.g. the web page is reloaded), another connection is made and a new watcher appears in kamailio without the old one disappearing. For example: # kamctl rpc presence.watcher_list basic sip:1001@example.voismart.com { "jsonrpc": "2.0", "result": [ { "pres_uri": "sip:1001@example.com", "to_user": "1001", "watcher_user": "1000", "contact": "sip:1000@mfp0h2dhhmq2.invalid;transport=ws;alias=192.168.56.10~34166~5", "callid": "1o9b499kj6l65if0p30t", "user_agent": "sip.js/1.12.1", "expires": 1736422494, ... }, { "pres_uri": "sip:1001@example.com", "to_user": "1001", "watcher_user": "1000", "contact": "sip:1000@0j7qoabnv827.invalid;transport=ws;alias=192.168.56.10~34156~5", "callid": "8mjhrqifgm39up9t2qq3", "user_agent": "sip.js/1.12.1", "expires": 1736422492, ... } ], "id": 11753 } you can see the duplicated watcher in which the old connection is now invalid because the websocket client disappeared. In this situation, issuing a pres_refresh_watchers() call will fail to send any notify (even to the valid connection) because of this error: WARNING: <core> [core/msg_translator.c:3007]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found ERROR: tm [t_msgbuilder.c:1423]: assemble_via(): via building failed ERROR: tm [t_msgbuilder.c:1614]: build_uac_req(): error while assembling Via ERROR: tm [uac.c:552]: t_uac_prepare(): Error while building message ERROR: presence [notify.c:1734]: send_notify_request(): in function tm t_request_within() ERROR: presence [notify.c:1829]: notify(): sending Notify not successful ERROR: presence [notify.c:1585]: query_db_notify(): Could not send notify for [event]=dialog ERROR: presence [presence.c:739]: pres_refresh_watchers(): sending Notify requests in which the building of the notify message fails because it cannot find a valid connection when building the via header. Is this a wanted behaviour? shouldn't it just fail the sending of the single message? Is there anything i can do to fix this? Trying to reproduce the same problem with a regular tcp client, it seems that the other notifys are correctly sent and only a timeout error is sent for the single message: ERROR: <core> [core/tcp_main.c:4622]: handle_tcpconn_ev(): connect 192.168.56.10:21994 failed which i think is the correct behaviour. Related question: is it possible to remove active watchers when the connection goes down? Thanks for the help on this.

2 5

Performance issues with KEMI
by Daniel W. Graham 20 Jan '25

20 Jan '25

Testing out KEMI functionality and running into performance issues. If I exceed 150 calls per second the network receive queue grows and Kamailio is unable to keep up with requests and they begin dropping. KEMI script for testing is just doing a stateless reply to invites. Using python3s module. I've played with Kamailio child processes and memory allocations, but there is no impact. I've also attempted some buffer / memory tweaking at the OS level, again with no impact. Increasing CPU cores and even running the test on bare metal results in the same. Example of receive queue at 150 calls per second - Netid State Recv-Q Send-Q Local Address:Port udp UNCONN 337280 0 x.x.x.x:5060 Just wondering if anyone has experienced similar issues or has an example of the performance they are seeing before I continue down this path. Thanks, - dan

5 9

Need help with serial branching using: t_load_contacts(1)
by Benoît Panizzon 20 Jan '25

20 Jan '25

Hi Gang I guess, I don't completely understand who to properly perform serial branching. I did try to follow the examples from: https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.f.t_next_con… This is, stripped down, the relevant config used. modparam("tm", "contacts_avp", "tm_contacts") modparam("tm", "contact_flows_avp", "tm_contact_flows") route[LOCATION] { lookup_to_dset("location", "$var(lookupuri)"); if (!t_load_contacts(1)) { xlog("L_ERR", "$cfg(route): ######### load_contacts failed\n"); sl_send_reply("500", "Server Internal Error - Cannot load contacts"); exit; } xlog("L_INFO", "$cfg(route): Contacts loaded: $cnt($xavp(tm_contacts))\n"); => Confirms, there is more than one contact loaded. if (!t_next_contacts()) { send_reply("480", "Not registered"); } t_set_fr(5000,1500); # Set 5 second timeout for LAB testing to quickly try the next contact. t_on_branch("BR_TO_CUST"); t_on_branch_failure("BR_TO_CUST_FAILURE"); exit; } event_route[tm:branch-failure:BR_TO_CUST_FAILURE] { xlog("BRANCH FAILED $T_reply_code to $rm message\n"); t_on_branch("BR_TO_CUST"); t_on_branch_failure("BR_TO_CUST_FAILURE"); if (t_next_contact_flow()) { xlog("BRANCH FAILED, Try Next\n"); t_relay(); } else { xlog("L_INFO", "No more flows\n"); t_reply("408", "Branch Timeout"); exit; } } What happens is: INVITE is sent to the first contact, who is replying RINGING. After 5 seconds the timeout is reached and the branch-failure route engaged. Jan 17 15:01:15 dev-cpereg01 kamailio[3599432]: ERROR: <script>: BRANCH FAILED 408 to INVITE message Jan 17 15:01:15 dev-cpereg01 kamailio[3599432]: INFO: <script>: No more flows Jan 17 15:01:15 dev-cpereg01 kamailio[3599432]: CRITICAL: tm [tm.c:1554]: ki_t_reply(): w_t_reply entered in unsupported mode To my understanding, t_next_contact_flow() should have loaded the next ds, but this does not seem to happen. What am I missing? -- 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 ______________________________________________________

2 3

Kamailio IMS, used with IPv6?
by Valentin Christoph 16 Jan '25

16 Jan '25

Anybody experiences with Kamailio IMS modules (ims_registrar_scscf, ims_usrloc_scscf, ims_isc, ........) and the use of IPv6 addresses? Any pitfalls? DNS issues? Thanks, Christoph

2 2

Asynchttp failover
by Sergio Charrua 15 Jan '25

15 Jan '25

Hello all! just wondering if anyone knows a good way to failover with the HTTP Async module. The standard HTTP Client module (synchronous) does implement some nice functionalities for failover between URL addresses, but the HTTP Async doesn't have any failover functionality... I have implemented some functions (within kamailio.cfg) for failover upon HTTP requests, but despite it working fine, it would be better to have that functionality on the HTTP Async module, just like HTTP Client has... Any technical reason for HTTP Async module not having failover functionalities? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

2 2

Module snmpstats reporting inconsistent call success metrics on intense peaks of failed calls
by Igor Potjevlesch 15 Jan '25

15 Jan '25

Hello! We are plotting the call success rate on each of our proxies. Using the Kamailio snmpstats module, we try to obtain three counters to plot them on a graph: Total Calls Active Calls Error Calls Most of the time, everything works well, and we obtain sensible values: snmpwalk -c mycommunity -v 2c W.X.Y.Z 1.3.6.1.4.1.34352.3.1.3.1.3.2 -O n .1.3.6.1.4.1.34352.3.1.3.1.3.2.1.0 = Gauge32: 309 -> Total Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.2.0 = Gauge32: 95 -> Active Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.3.0 = Gauge32: 214 -> Error calls However, when our proxies are facing intense peaks of failed calls in a short time span, sometimes, the "Error Calls" counter returns a value higher than "Total Calls". As a result, "Active Calls" suddenly reaches a value close to the maximum of a Gauge32 (because T o t a l C a l l s − E r r o r C a l l s < 0 ). snmpwalk -c mycommunity -v 2c W.X.Y.Z 1.3.6.1.4.1.34352.3.1.3.1.3.2 -O n .1.3.6.1.4.1.34352.3.1.3.1.3.2.1.0 = Gauge32: 153 -> Total Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.2.0 = Gauge32: 4294967226 -> Active Calls .1.3.6.1.4.1.34352.3.1.3.1.3.2.3.0 = Gauge32: 223 -> Error calls As a result, our graphs make no more sense. We are currently running kamailio 5.6.4. Any idea if it's a bug or a configuration issue? Regards, Igor. -- Cet e-mail a été vérifié par le logiciel antivirus d'Avast. www.avast.com

3 7

Branching 6.0 series
by Daniel-Constantin Mierla 15 Jan '25

15 Jan '25

Hello, I am trying to figure out what would be the best date to branch 6.0 in the git repo. To allow a bit of time after returning from the winter holidays to tune more the cmake and packaging, I would think that January 15, 2025 could be a good choice. If nobody comes with other opinions/suggestions, then we will go for it. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

kamcmd tls.reload issue
by Yuriy Nasida 14 Jan '25

14 Jan '25

Hello, I am using letsencrypt cert and key and do not want to restart kamailio every 3 months to load new ones. I know that there is: kamcmd tls.reload method but it has an error for me. error: 500 - Error while fixing TLS configuration (consult server log) I am checking the logs and see: kamailio[3865480]: INFO: tls [tls_domain.c:345]: ksr_tls_fill_missing(): TLSs<default>: tls_method=3 kamailio[3865480]: INFO: tls [tls_domain.c:357]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/kamailio/certs/my_cert.crt' kamailio[3865480]: INFO: tls [tls_domain.c:364]: ksr_tls_fill_missing(): TLSs<default>: ca_list='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:371]: ksr_tls_fill_missing(): TLSs<default>: ca_path='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:378]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:382]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 kamailio[3865480]: INFO: tls [tls_domain.c:390]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' kamailio[3865480]: INFO: tls [tls_domain.c:397]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/kamailio/certs/private.key' kamailio[3865480]: INFO: tls [tls_domain.c:401]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 kamailio[3865480]: INFO: tls [tls_domain.c:406]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 kamailio[3865480]: INFO: tls [tls_domain.c:410]: ksr_tls_fill_missing(): TLSs<default>: verify_client=0 kamailio[3865480]: NOTICE: tls [tls_domain.c:1168]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... kamailio[3865480]: ERROR: tls [tls_domain.c:590]: load_cert(): TLSs<default>: Unable to load certificate file '/etc/kamailio/certs/my_cert.crt' kamailio[3865480]: ERROR: tls [tls_util.h:49]: tls_err_ret(): load_cert:error:03000072:digital envelope routines::decode error (sni: unknown) kamailio[3865480]: ERROR: tls [tls_util.h:49]: tls_err_ret(): load_cert:error:0A00018F:SSL routines::ee key too small (sni: unknown) Any advice ? It's interesting that there are not any errors in case I restart kamailio. I can make TLS calls without problems. deb 12.5 version: kamailio 5.7.4 (x86_64/linux)

4 5

FOSDEM VoIP Dinner
by Torrey Searle 14 Jan '25

14 Jan '25

Hello all, As in previous years, I'm organising the VoIP dinner for FOSDEM (1 Feb @ 19:00) However unlike previous years (since the restaurant got upset about splitting the bill 30 ways). I will ask everyone to order their meal and pay in advance. (and I will pay the bill to the restaurant with my card) If you are wanting to attend the dinner please fill out the following form and follow the instructions for payment. https://form.jotform.com/250012543310033 Contact details are also in the form if you want to reach out to me with any further questions. Kind regards Torrey

5 4

ims_diameter_server not freeing shared memory
by marpett365＠gmail.com 09 Jan '25

09 Jan '25

Hi, I have run in to an issue when sending diameter requests using the ims_diameter_server module. Shared memory keeps increasing and are not being freed until restart of kamailio. This leads to all shared memory is being used and no more are able to be allocated. This only happens when the request is sent synchronously. When sent asynchronously shared memory is being freed as far as I can see. have anyone ran into this issue and found a remedy or can give some pointers how to proceed? Following functions have been used for sending requests. diameter_request([peer], appid, commandcode, message) diameter_request_async([peer], appid, commandcode, message) route[CHECK_LOCATION] { diameter_request("hss.ims.test.local", "16777217", "306", "[ " + "{ \"avpCode\":260, \"vendorId\":0, \"Flags\":64, \"list\":[ { \"avpCode\":266, \"vendorId\":0, \"Flags\":64, \"int32\":10415 }, { \"avpCode\":258, \"vendorId\":0, \"Flags\":64, \"int32\":16777217 } ] }, " + "{ \"avpCode\":277, \"vendorId\":0, \"Flags\":64, \"int32\":1 }, " + "{ \"avpCode\":283, \"vendorId\":0, \"Flags\":64, \"string\":\"ims.test.local\"}, " + "{ \"avpCode\":706, \"vendorId\":10415, \"Flags\":192, \"int32\":0 }, " + "{ \"avpCode\":700, \"vendorId\":10415, \"Flags\":192, \"list\": [ { \"avpCode\":601, \"vendorId\":10415, \"Flags\":192, \"string\":\"tel:" + $fU + "\" }]}, " + "{ \"avpCode\":707, \"vendorId\":10415, \"Flags\":192, \"int32\":0 }, " + "{ \"avpCode\":703, \"vendorId\":10415, \"Flags\":192, \"int32\":14 } ]"); xlog("L_INFO", "Sent Diameter request"); } Shared memory after restart kamcmd mod.stats cdp shm Module: cdp { save_peer_applications(779): 144 make_send_pipe(126): 320 dp_add_pid(92): 216 AAAAddRequestHandler(69): 40 cb_add(183): 8 cb_add(177): 32 cdp_sessions_init(178): 8 cdp_sessions_init(173): 8 cdp_sessions_init(165): 1024 cdp_sessions_init(157): 3072 cdp_sessions_init(149): 8 cdp_trans_init(67): 8 cdp_trans_init(60): 24 add_timer(150): 24 add_timer(145): 144 new_peer(77): 16 new_peer(75): 16 new_peer(72): 16 new_peer(69): 80 new_peer(66): 96 new_peer(60): 384 peer_manager_init(84): 8 peer_manager_init(83): 8 peer_manager_init(82): 8 peer_manager_init(79): 8 peer_manager_init(76): 16 worker_init(101): 16 worker_init(94): 128 worker_init(89): 32 worker_init(87): 32 worker_init(84): 8 worker_init(82): 48 timer_cdp_init(182): 8 timer_cdp_init(179): 16 diameter_peer_init_real(137): 8 diameter_peer_init_real(131): 16 diameter_peer_init_real(122): 16 diameter_peer_init_real(115): 8 diameter_peer_init_real(108): 8 diameter_peer_init_real(101): 8 diameter_peer_init_real(94): 8 new_routing_entry(88): 64 parse_dp_config(447): 16 parse_dp_config(348): 8 parse_dp_config(340): 16 parse_dp_config(333): 40 parse_dp_config(326): 144 quote_trim_dup(96): 456 new_dp_config(52): 184 Total: 7024 } Shared memory after 10 diameter requests Module: cdp { save_peer_applications(779): 144 AAACreateAVP(137): 5072 AAATranslateMessage(500): 1320 do_receive(600): 7392 make_send_pipe(126): 320 dp_add_pid(92): 216 AAAAddRequestHandler(69): 40 cb_add(183): 8 cb_add(177): 32 cdp_sessions_init(178): 8 cdp_sessions_init(173): 8 cdp_sessions_init(165): 1024 cdp_sessions_init(157): 3072 cdp_sessions_init(149): 8 cdp_trans_init(67): 8 cdp_trans_init(60): 24 add_timer(150): 24 add_timer(145): 144 new_peer(77): 16 new_peer(75): 16 new_peer(72): 16 new_peer(69): 80 new_peer(66): 96 new_peer(60): 384 peer_manager_init(84): 8 peer_manager_init(83): 8 peer_manager_init(82): 8 peer_manager_init(79): 8 peer_manager_init(76): 16 worker_init(101): 16 worker_init(94): 128 worker_init(89): 32 worker_init(87): 32 worker_init(84): 8 worker_init(82): 48 timer_cdp_init(182): 8 timer_cdp_init(179): 16 diameter_peer_init_real(137): 8 diameter_peer_init_real(131): 16 diameter_peer_init_real(122): 16 diameter_peer_init_real(115): 8 diameter_peer_init_real(108): 8 diameter_peer_init_real(101): 8 diameter_peer_init_real(94): 8 new_routing_entry(88): 64 parse_dp_config(447): 16 parse_dp_config(348): 8 parse_dp_config(340): 16 parse_dp_config(333): 40 parse_dp_config(326): 144 quote_trim_dup(96): 456 new_dp_config(52): 184 Total: 20808 } Shared memory after 170k requests do_receive(600): 131363752 AAATranslateMessage(500): 32326656 AAACreateAVP(137): 110067152 Total: 273764840 Br Martin

2 5

evapi not processing received messages
by Sergio Charrua 09 Jan '25

09 Jan '25

Hi all! I have implemented evapi module on Kamailio 5.8.4, calling a python script. Python script is listening to a socket. It receives messages (JSON) from Kamailio, makes an HTTP request somewhere, and sends the response back to Kamailio. The Python script is working fine: it connects to Kamailio, sends HTTP requests, and forwards a response back to Kamailio: I have confirmed using TCPDump that, in fact, the data is received and forwarded back. However, despite showing messages that the Python client script is correctly connected to Kamailio, the event_route[evapi:message-received] is never executed. Probably there is some detail that I am missing, but the docs and online info for evapi are very rare, and they all show very similar configuration. And as I wrote, I have confirmed using TCPDump that the data is received from Python client script to Kamailio node. What am I missing? This is my Kamailio cfg: [loads modules] [sets modparams] modparam("evapi","bind_addr","10.20.20.1:8888") debug=2 children=16 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " disable_sctp = yes force_rport = yes rundir="/tmp" request_route { if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); } exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } route(TOEVAPI); exit; } event_route[evapi:connection-new] { xlog("LINFO","new connection from [$evapi(srcaddr):$evapi(srcport)]\n"); if($evapi(srcaddr)!="10.20.0.1") { evapi_close(); exit; } } event_route[evapi:connection-closed] { xlog("LINFO","connection closed by $evapi(srcaddr):$evapi(srcport)\n"); } event_route[evapi:message-received] { xlog("LINFO","received [$evapi(msg)] from $evapi(srcaddr):$evapi(srcport)\n"); jansson_get("t-index", "$evapi(msg)", "$var(t-index)"); jansson_get("t-label", "$evapi(msg)", "$var(t-label)"); $var(evmsg) = $evapi(msg); xlog("L_INFO", "preparing to resume transaction for processing: $var(tindex) / $var(tlabel)\n"); t_continue("$var(tindex)", "$var(tlabel)", "EVAPIRESPONSE"); } route[EVAPIRESPONSE] { xlog("L_INFO", "resumed transaction for processing: $T(id_index) / $T(id_label)\n"); [do stuff] } route[TOEVAPI]{ xlog("L_INFO", "suspended transaction: $T(id_index) / $T(id_label)\n"); evapi_async_relay( [some JSON string] ); exit; } Thanks in advance! Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 6

Kamailio and Faxing
by Temilolu Fayomi 09 Jan '25

09 Jan '25

Hello, I am having some issues ensuring t38 faxes are being received reliably. I am currently using kamailio as an inbound sbc with freepbx as the endpoint. I understand that potentially stripping other codecs except for g711a/u makes inbound faxes reliable, but beyond that I am having trouble determining how to ensure reliable faxing. Is this something kamailio can handle (such as an error correction mode or pass through setting or module?) I would appreciate any insight as I have hit a wall when it comes to this issue. Thank you, Temi

3 3

Re: WS and rtpengine_manage
by Knserbrave 08 Jan '25

08 Jan '25

test

2 4

deb-archive.kamailio.org unreachable
by Floimair Florian 08 Jan '25

08 Jan '25

Hi! It seems that deb-archive.kamailio.org is currently down. It resolves to IP 76.9.245.162 on my system Can someone with access to that please have a look. Thank you FLORIAN FLOIMAIR Software Development - Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria [signature_2459749227] commend.com LG Salzburg / FN 178618z

2 2

Kamailio crash
by Mitesh Thakkar 07 Jan '25

07 Jan '25

Hi, I have recently migrated from Kamailio 5.2.5 to Kamailio 5.5.7. After a migration, the Kamailio crashed. Find the details below. Thanks in advance. Back trace: ========= Core was generated by `/usr/local/sbin/kamailio -DD -E -m 1941 -M 970 -f /etc/kamailio/kamailio.cfg --'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f8e27a2be15 in OPENSSL_cleanse () from /lib/x86_64-linux-gnu/libcrypto.so.3 (gdb) bt #0 0x00007f8e27a2be15 in OPENSSL_cleanse () from /lib/x86_64-linux-gnu/libcrypto.so.3 #1 0x00007f8e278faa8a in CRYPTO_clear_free () from /lib/x86_64-linux-gnu/libcrypto.so.3 #2 0x00007f8e27838175 in BUF_MEM_free () from /lib/x86_64-linux-gnu/libcrypto.so.3 #3 0x00007f8e27bc0c98 in SSL_free () from /lib/x86_64-linux-gnu/libssl.so.3 #4 0x00007f8e27c8cd6e in tls_h_tcpconn_clean_f (c=0x7f8db0fa1e90) at tls_server.c:701 #5 0x000055f7f42cc6e1 in _tcpconn_free (c=0x7f8db0fa1e90) at core/tcp_main.c:1541 #6 0x000055f7f42e2f3c in tcpconn_put_destroy (tcpconn=0x7f8db0fa1e90) at core/tcp_main.c:3285 #7 0x000055f7f42e8759 in handle_tcp_child (tcp_c=0x7f8e27d88078, fd_i=-1) at core/tcp_main.c:3722 #8 0x000055f7f42f506f in handle_io (fm=0x7f8e27db1578, ev=1, idx=-1) at core/tcp_main.c:4539 #9 0x000055f7f42b6e49 in io_wait_loop_epoll (h=0x55f7f45ae240 <io_h>, t=5, repeat=0) at core/io_wait.h:1070 #10 0x000055f7f42f8bc3 in tcp_main_loop () at core/tcp_main.c:4824 #11 0x000055f7f4057fc2 in main_loop () at main.c:1856 #12 0x000055f7f4062670 in main (argc=10, argv=0x7ffc6d583528) at main.c:3052 OS: Ubuntu 22.04.5 LTS \n \l # kamailio --version version: kamailio 5.5.7 (x86_64/linux) 272fdd 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: 272fdd compiled on 14:21:35 Oct 21 2024 with gcc 11.4.0 # dpkg -l |grep openssl ii openssl 3.0.2-0ubuntu1.18 amd64 Secure Sockets Layer toolkit - cryptographic utility Regards.

3 3

children monitoring
by Sergey Safarov 06 Jan '25

06 Jan '25

If TCP children are busy, Please correct me if I am wrong if the remote IP is not responding via TCP protocol, this leads to a timeout and this child is busy. If many remote endpoints are not reachable via TCP this can lead to the case when half or more children are by timeout operations. I think required to monitor how many children are busy. Is it possible to get such info using kamcmd utility, statistics module, or xprom module. Sergey

4 7

Crash on dialog_publish.
by satyaprakash ch 06 Jan '25

06 Jan '25

Hi, The kamailio version we are using is "5.7.4", We are facing a kamailio crash recently and it has generated the core-dump. Below is the backtrace for the coredump. Could you please assist with this crash. *Bachtrace:* #0 0x00007fa6b051748a in dialog_publish (state=0x7fa6b0519505 "terminated", ruri=0x7069734030783532, entity=0x7ffcdfb16510, peer=0x7ffcdfb16520, callid=0x7fa5b0527cb8, initiator=1, lifetime=10, localtag=0x0, remotetag=0x0, localtarget=0x7fa5b0527ce8, remotetarget=0x7ffcdfb16500, do_pubruri_localcheck=0, uuid=0x7fa5b0527d18) at dialog_publish.c:302 body = 0x0 uri = {s = 0x0, len = 0} publ = 0x0 size = 0 content_type = {s = 0x0, len = -1288318368} ruri_uri = {user = {s = 0x0, len = 0}, passwd = {s = 0x0, len = 0}, host = {s = 0x0, len = 0}, port = {s = 0x0, len = 0}, params = {s = 0x0, len = 0}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = (unknown: 0), transport = {s = 0x0, len = 0}, ttl = {s = 0x0, len = 0}, user_param = {s = 0x0, len = 0}, maddr = {s = 0x0, len = 0}, method = {s = 0x0, len = 0}, lr = {s = 0x0, len = 0}, r2 = {s = 0x0, len = 0}, gr = {s = 0x0, len = 0}, transport_val = {s = 0x0, len = 0}, ttl_val = {s = 0x0, len = 0}, user_param_val = {s = 0x0, len = 0}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 0}, lr_val = {s = 0x0, len = 0}, r2_val = {s = 0x0, len = 0}, gr_val = {s = 0x0, len = 0}} __func__ = "dialog_publish" #1 0x00007fa6b0518f89 in dialog_publish_multi (state=0x7fa6b0519505 "terminated", ruris=0x7069734030783532, entity=0x7ffcdfb16510, peer=0x7ffcdfb16520, callid=0x7fa5b0527cb8, initiator=1, lifetime=10, localtag=0x0, remotetag=0x0, localtarget=0x7fa5b0527ce8, remotetarget=0x7ffcdfb16500, do_pubruri_localcheck=0, uuid=0x7fa5b0527d18) at dialog_publish.c:412 __func__ = "dialog_publish_multi" #2 0x00007fa6b0506944 in __dialog_sendpublish (dlg=0x7fa5b046a850, type=64, _params=0x7fa6b0437620 <params>) at pua_dialoginfo.c:402 tag = {s = 0x0, len = 0} uri = { s = 0x7fa5b0527d41 "sip:+1248XXXXXXX@10.XXX.X.Xm6ka0542f9v0lmjjc4npkamailio954739148rdb2847sip :XXXXXX.com:5060sip:521XXXx0;tgrp=51;trunk-context=198.X.XXX.XXX@10.XXX.XX.XX:5060padi-6761ad46-154e-1560padi\300\300\300\300", len = 29} identity_local = { s = 0x7fa5b0527d28 "sip:413XXXx0@10.XXX.X.XXsip :+1248XXXXXXXX@10.XXX.X.XXm6ka0542f9v0lmjjc4npkamailio954739148rdb2847sip :XXXXXX.com:5060sip:415XXXx0;tgrp=51;trunk-context=198.X.XXX.XXX@10.XXX.XX.XX:5060padi-6761ad46-154e-"..., len = 25} target = {s = 0x0, len = 0} dlginfo = 0x7fa5b0527c90 request = 0x7fa6b3361b58 __func__ = "__dialog_sendpublish" Thanks.

1 2

record_route() adding wrong sips URI when transport is missing (defaulting to UDP)
by Benoit Panizzon 06 Jan '25

06 Jan '25

Hi Gang A quick search via google did not reveal any useful hits. Kamailio 5.7.6 CPE (TLS) => SBC (UDP) => Registrar (UDP) => Core CPE uses a sips contact URI to register, no transport attribute! sips:315996608@172.29.255.11:5061 SBC uses Path functionality. Situation: Call to CPE I see the registrar adding this Record-Route: Record-Route: <sips:registrar-ip;lr;ftag=6159862378> The SBC is adding a r2=on Record-Route to indicate the change between UDP and TLS. The Route-Set received by the CPE then is: INVITE sips:315996608@172.29.255.11:5061 SIP/2.0 Record-Route: <sips:sbc-ip:5061;transport=tls;r2=on;lr;ftag=6159862378> Record-Route: <sips:sbc-ip;r2=on;lr;ftag=6159862378> Record-Route: <sips:registrar-ip;lr;ftag=6159862378> Record-Route: <sip:core-ip;lr;ftag=6159862378> 180 Ringing req 100rel is making it back as the Via header look right. But when the remote side is sending a PRACK, is has to follow the Routeset which if I am not mistaking, is not making it back because because the CORE does not know how to send this via TLS to the Registrar. Core logging: prepare_new_uac(): can't fwd to af 2, proto 3 (no corresponding listening socket) IMHO already the 2nd Record-Route added by the Registrar should not contain a sips: URI but a sip: URI (reflecting the transport attribute not being present, defaulting to UDP) Is this a known issue? 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 4

Guidance Required for Kamailio Load Balancer Behavior in Attended Transfer Scenarios
by Kotappa Gandudi 03 Jan '25

03 Jan '25

Dear Kamailio Team, I hope this email finds you well. I am currently working on a project where Kamailio is being utilized as a load balancer to distribute calls across multiple FreeSWITCH instances. I am using the round-robin technique for load distribution, and it is working perfectly for regular call scenarios. However, I am encountering an issue when handling attended transfer calls. Specifically, for a normal call, Kamailio assigns one FreeSWITCH instance, but during the attended transfer call, it assigns another FreeSWITCH instance. This behavior disrupts the continuity of the call handling. My requirement is for Kamailio to use the same FreeSWITCH instance for the entire call, including any attended transfers. I believe maintaining the same FreeSWITCH instance for the entire call flow would ensure consistency and avoid potential issues. Could you please guide me on how to configure Kamailio to achieve this behavior? I would greatly appreciate any suggestions, configurations, or modules that can help resolve this issue. Thank you for your time and assistance. Looking forward to your response. Best regards, Kotappa Y Gandudi

4 4

how to call loose_route() after adding route header without using msg_apply_changes?
by Benoît Panizzon 31 Dec '24

31 Dec '24

Hi List To handle CPE which can only handle a limited number of route header (PacketCable SIP specification), I need to store and restore the route header (limited topology hiding only affecting Route Header) to present the CPE only one Route or RR header. I pass the ID of the stored route-set via RR param. So when I get a message back from the CPE is: * Take ID from RR param and pull stored route set from cache. * remove_hf("Route"); * insert_hf("Route: $avp(stored-routeset)\r\n"); * msg_apply_changes(); * loose_route(); I found loose_route() only to be able to correctly determine the next hop, if I first did apply the changes. But after adding rtpengine to the mix, I got back into the known situation that calling rtpengine (on the branch route) after calling msg_apply_changes() results in duplicate SDP. Is there a way to get loose_route() to operate on freshly added header without calling msg_apply_changes? -- 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 ______________________________________________________

8 28

Async HTTP request is sent but response is incorrectly processed
by Sergio Charrua 30 Dec '24

30 Dec '24

Hi all! still struggling with Async HTTP Client ... My current configuration is very simple: #!define HTTP_API_ROUTING_ENDPOINT_MAIN " http://MY_HTTP_REST_API:18085/Endpoint" include_file "listen.cfg" loadmodule "pv" loadmodule "tm" loadmodule "http_async_client" loadmodule "xlog" loadmodule "tmx" loadmodule "sl" loadmodule "textops" loadmodule "cfgutils" loadmodule "siputils" loadmodule "rr" loadmodule "ctl" loadmodule "jsonrpcs.so" loadmodule "jansson.so" modparam("pv", "shvset", "http_async=i:8"); modparam("http_async_client", "workers", 1); modparam("http_async_client", "connection_timeout", 500) modparam("http_async_client", "curl_verbose", 1) # for debug purpose #modparam("tm", "fr_timer", 750) #modparam("tm", "fr_inv_timer", 20000) #modparam("tm", "remap_503_500", 0) modparam("tm", "failure_reply_mode", 3) modparam("tm", "max_inv_lifetime",5000) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") debug=2 children=16 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " disable_sctp = yes force_rport = yes rundir="/tmp" request_route { $avp(REQUEST_URL) = HTTP_API_ROUTING_ENDPOINT_MAIN; if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); } exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } jansson_set("string","from" , $hdr(From) , "$var(req)"); jansson_set("string","to" , $hdr(To) , "$var(req)"); jansson_set("string","r-uri" , $ru , "$var(req)"); jansson_set("string","contact" , $hdr(Contact) , "$var(req)"); jansson_set("string","call-id" , $ci , "$var(req)"); xinfo("MAIN-ROUTE - Setting From=$hdr(From) To=$hdr(To) Contact=$hdr(Contact) ci=$ci \n"); if ( is_present_hf("Identity") ) jansson_set("string","identity" , $hdr(Identity) , "$var(req)"); if ( is_present_hf("P-Identity-Bypass") ) jansson_set("string","p-identity-bypass" , $hdr(P-Identity-Bypass) , "$var(req)"); if ( is_present_hf("P-Asserted-Identity") ) jansson_set("string","p-asserted-identity" , $hdr(P-Asserted-Identity) , "$var(req)"); if ( is_present_hf("P-STSH-UC") ) jansson_set("string","p-stsh-uc" , $hdr(P-STSH-UC) , "$var(req)"); jansson_set("string", "request-time" , $avp(requestTime), "$var(req)"); ## Create transaction before http_async_query; if (is_method("INVITE")){ t_newtran(); xinfo("MAIN-ROUTE - Sending request [$var(req)]\n"); route(HTTP_ASYNC_REQUEST); } exit; } route[HTTP_ASYNC_REQUEST] { xinfo("HTTP_ASYNC_REQUEST - Sending request [$var(req)]\n"); $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = 1500; # 1 second timeout $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; $http_req(body) = $var(req); if ( !http_async_query($avp(REQUEST_URL), "HTTP_REPLY") ) { xinfo("HTTP_ASYNC_REQUEST - http_async_query $ci FAILED. Return is [$rc]\n"); }else{ send_reply("503", "Server Error"); exit; } } route[HTTP_REPLY] { xinfo("HTTP_REPLY - Xphere Response Code $http_rs - $ci\n"); if ( $http_ok && $http_rs==200) { xinfo("HTTP_REPLY - $ci reply ok\n"); jansson_xdecode($http_rb, "json"); xinfo("HTTP_REPLY - HTTP JSON Response: $http_rb \n"); if ( $xavp(json=>sip-response-code) == 300 ){ remove_hf("Contact"); append_to_reply("Contact: $xavp(json=>contact)\r\n"); ## "touch" the ruri to add the contact header for 3xx: $ru = $ru; send_reply("300", "Multiple Choice"); exit; }else{ xinfo("HTTP_REPLY - $ci Error Replied $xavp(json=>sip-response-code) - $xavp(json=>sip-response-text) \n"); send_reply("$xavp(json=>sip-response-code)","$xavp(json=>sip-response-text)"); exit; } } else { xinfo("HTTP_REPLY - HTTP Response Code $ci - $http_rs and error $http_err \n"); if (!$http_ok){ sl_reply("503", "No Connection to HTTP"); exit; }else if ($http_rs>=500){ sl_reply("503", "HTTP Processing Error"); exit; } else{ sl_reply("503", "Cannot Connect to HTTP"); exit; } } exit; } The configuration is based on Ben Kaufman's kamailio_http_async <https://github.com/whosgonna/kamailio_http_async> example, but I have extended it a bit. Currently all the requests are being sent to the HTTP REST API. I can't find queue/buffering issues... What I am experiencing is that , using SNGREP, I get this flow: 10.240.12.209:5060 10.242.17.146:5060 ----------*---------- ----------*---------- 17:33:46.689852 x INVITE (SDP) x +0.000530 x --------------------------> x 17:33:46.690382 x 100 trying -- your call is x +1.499822 x <-------------------------- x 17:33:48.190204 x 503 No Connection to HTTP x +0.000029 x <-------------------------- x 17:33:48.190233 x 500 I'm terribly sorry, se x +0.002130 x <-------------------------- x 17:33:48.192363 x ACK x +0.485836 x --------------------------> x 17:33:48.678199 x 500 I'm terribly sorry, se x +0.004881 x <<<------------------------ x 17:33:48.683080 x ACK x +0.973295 x ------------------------>>> x 17:33:49.656375 x 500 I'm terribly sorry, se x +0.004105 x <<<------------------------ x 17:33:49.660480 x ACK x +1.995420 x ------------------------>>> x 17:33:51.655900 x 500 I'm terribly sorry, se x +0.005361 x <<<-------------------------x 17:33:51.661261 x ACK x x ------------------------>>> x I suspected issues with the HTTP REST service, but when using tcpdump, I can find the Request from Kamailio and a reply back from the REST API service, so doesn't seem to be a problem on the HTTP REST service So my questions are: - why is Kamailio sending a "SIP 500 I'm terribly sorry..." error after a 503 No Connection to HTTP is sent and an "exit" command is executed? - how can I get rid of these messages? - Why Kamailio keeps sending responses back to SBC after the ACK when it is set to just "exit"? - Why is Kamailio not processing the response from HTTP REST API when it is correctly received? Logs are also weird: - I find a lot of these warnings: "Dec 29 17:38:34 ire-nprod-sip1 /usr/local/sbin/kamailio[590864]: WARNING: http_async_client [http_async_client_mod.c:529]: ah_get_status(): an async variable was read after http error, use $http_ok to check the request's status" and "Dec 29 17:38:34 ire-nprod-sip1 /usr/local/sbin/kamailio[590864]: ERROR: http_async_client [http_multi.c:621]: check_multi_info(): handle 0x7f375f786b30 returned error 7:" (I am still not using EVAPI module, that will be my next step, after the holiday season) (I suspect, however, that the REST API is not being able to handle the load in this test environment (between 400 to 600 CAPS), but that is another matter.) In attach, TCPDUMP, if anyone wishes to analyse. Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 5

http sync / async final tests
by Alexis Fidalgo 26 Dec '24

26 Dec '24

Just to share, and thanks for your valuable help. My conclusion after tests (on my scenario) is http backend: is a golang http server with a 250ms delay to answer, and answering back a json, running in 10 replicas inside a K8 with 4 nodes. test (from kamailio perspective): receive INVITE send 100 call the http, get the result, get a key from the result and set it as the value of Contact header send 302 receive ACK (discard on sync scenario because it’s stateless, use it in async to end transaction) Kamailio host: 1vcpu kamailio 3.8.4 bookworm running inside containerd (with podman) 16GB RAM sipp: run from the K8 nodes hosts (directly from OS) (everything is LAN connected on 2 different physical servers, hypervisor is VMWare i guess 7.0.3) using http_client (sync with stateless): 8 children -M8 -m64 no more than 60 calls/sec (15cps on each sipp instance) until sipp start to retransmit the INVITES using async_http_client: 8 children -M8 -m2048 it’s been ~36 hours now at 250cps each sipp instance with 0 errors 0 retransmission and the kamailio host is userlab@sblb1:~$ vmstat 1 procs -----------memory---------- ---swap-- -----io---- -system-- ------cpu----- r b swpd free buff cache si so bi bo in cs us sy id wa st 4 0 0 10285460 364684 5444240 0 0 0 22 36 68 0 0 100 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5284 6822 27 14 58 0 0 4 0 0 10285460 364684 5444240 0 0 0 0 5306 6752 29 15 57 0 0 3 0 0 10285460 364684 5444240 0 0 0 0 5304 6823 26 16 57 0 0 2 0 0 10285460 364684 5444240 0 0 0 12 5287 6761 26 16 57 0 0 2 0 0 10285460 364684 5444240 0 0 0 0 5337 6838 27 14 58 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5101 6724 25 17 59 0 0 3 0 0 10285460 364684 5444240 0 0 0 0 5066 6650 26 14 60 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5306 6829 23 17 59 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5219 6725 23 19 58 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5274 6770 26 13 61 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5295 6756 26 18 56 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5235 6786 22 16 62 0 0 1 0 0 10285460 364684 5444240 0 0 0 0 5315 6833 21 20 59 0 0 at least for my scenario, it worths to pay the price of the memory tradeoff. again, Ben, Alex, Sergio, all, thanks for your help. regards

1 0

help on how to get ACK
by Alexis Fidalgo 24 Dec '24

24 Dec '24

Hello all, moving just a bit aside of the http and async_http. After all the real useful and interesting thread on that topic what helped me, im facing a problem i cant deal with and need a hint at least. Scenario INVITE -> Kamailio on request_route ... if(is_method("INVITE")){ t_newtran(); http_async_query("http://nuc:8080", "HTTP_REPLY"); } … Kamailio -> 100 - Trying then route[HTTP_REPLY] { if ($http_ok) { xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n"); t_reply(302,"Redirect"); } else { xlog("L_INFO", "route[HTTP_REPLY]: error $http_err)\n"); } } Kamailio -> 302 Redirect ACK -> Kamailio This last ACK, how can i read it and use it to terminate the transaction? because Kamailio keeps transmitting the 302 message 3 more times until the transaction is finished by a timer 42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted. buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3 42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted. buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3 42(44) DEBUG: tm [t_reply.c:1723]: t_retransmit_reply(): reply retransmitted. buf=0x7f4c44f9d680: SIP/2.0 3..., shmem=0x7f4c3fce7900: SIP/2.0 3 42(44) DEBUG: tm [timer.c:642]: wait_handler(): finished transaction: 0x7f4c3fcd35a0 (p:0x7f4c3fad85d0/n:0x7f4c3fad85d0) 42(44) DEBUG: tm [h_table.c:133]: free_cell_helper(): freeing transaction 0x7f4c3fcd35a0 from timer.c:651 in request_route i have if(is_method("ACK")){ if(!t_check_trans()){ t_release(); } exit; }

4 26

append_branch
by Alexis Fidalgo 24 Dec '24

24 Dec '24

Hello all, how can avoid to append_branch do this call to call SIP/2.0 302 Redirect Via: SIP/2.0/UDP 192.168.86.250:5060;branch=z9hG4bK-65665-5-0 From: sipp <sip:sipp@192.168.86.250:5060>;tag=65665SIPpTag005 To: 123456 <sip:123456@192.168.86.128:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-deb8da2e Call-ID: 5-65665(a)192.168.86.250 CSeq: 1 INVITE Contact: <sip:4093823670@otherdomain.com>, <sip:8888576523@otherdomain.com>, <sip:9719533197@otherdomain.com>, <sip:5302579529@otherdomain.com>, <sip:4972010218@otherdomain.com> Server: kamailio (5.8.4 (x86_64/linux)) Content-Length: 0 What i mean with “avoid”, call to call, contact should have only 1 branch, no a stack of previous. thanks

2 1

Transactions help
by Alexis Fidalgo 24 Dec '24

24 Dec '24

Hello, after the discussion regarding http and http_async, I started some labs to make a comparison and see what’s the trade off and the associated cost. Main problem (to me) is im not clear on how transactions behave. Im running some fast tests now before I start to deploy the final scenarios and 1. A new transaction is created before the http_async call (as the module example shows for http_async_query function 2. The http_async_query call is executed, receives a response and the HTTP_REPLY route for http_async_query is executed. 3. As we need to redirect the call, im adding Contact information with append_to_reply and then calling t_reply(302,”Redirect) Up to here INVITE is received, a 100 is answered and then the 302 with the expected Contact information (collected and built from the http_async_query response), problem is that the 302 is retransmitted 3 more times. So, im not even sure that im doing the correct actions to process the call this way. Is there a document/book/reference I can read to understand transactions better? By now I can not handle the call to be processed correctly (without the retransmissions) and finish the transaction the correct way. Any help, link, guide, will be appreciated. Regards.

6 17

Kamailio not receiving packets on high CPS
by Sergio Charrua 23 Dec '24

23 Dec '24

Hello! for this ST/SH project, we are using Kamailio 5.8.4 in stateless mode, making HTTP requests to a REST API (Java Spring) that will reply with a JSON object, and then Kamailio replaces the Contact header and sends a SIP 300 Multiple Choice reply to the SBC that sent the initial INVITE. Kamailio is running on a VM with 4vCPUs and 4GB RAM and 2 NICs. On NIC ens224 there is a Virtual IP managed by Keepalived. Kamailio is listening on port 5060 on ens224 and set with 32 children process (tried with 8, 16, 24 and also 64, but with this value it was a lot worst). We are doing load performance tests on our PreProd environment. With the SBC we are sending 450 to 600 CAPS to Kamailio, and what we noticed is that above 450 CAPS, and after less than 1 minute, Kamailio only replies SIP 100 to some INVITEs. We also could figure out that Kamailio is not receiving all the INVITES, despite having proved that ALL invites are sent to the server. It seems to me that, for some reason, the OS is somehow not able to deliver *all* SIP packets to Kamailio, because: - sngrep does capture all the SIP packets and shows the flows - though the flow shows SIP invite coming from SBC to Kamailio Server, there is no SIP 100 replied back to SBC, even though this is the 1st thing kamailio does, so, for me, Kamailio did not received the SIP Invite - picking a call with reinvites, which after some seconds, SBC cancels, if I copy the Call-ID value and grep on kamailio's logs, I just do not find any content! I have read some "network tuning" articles, including Linux Tune Network Stack (Buffers Size) To Increase Networking Performance - nixCraft <https://www.cyberciti.biz/faq/linux-tcp-tuning/> and Tuning Kamailio for high throughput and performance | Evariste Systems Blog <https://blog.evaristesys.com/2016/02/15/tuning-kamailio-for-high-throughput…> and when running the following command I get: $ ss -4 -n -l | grep 5060 udp UNCONN 0 0 10.242.17.125:5060 0.0.0.0:* udp UNCONN 324224 0 10.242.17.146:5060 0.0.0.0:* According to the article, the 3rd column should be as near 0 as possible, ideally 0. While I am waiting for some sysadmin with root permissions to be available and modify some of the network parameters on this VM, I wonder if anyone has some tips to share on how to solve this behaviour. I'm not sure, either, what parameters to change or if they should be changed! The goal is to have a minimum 750 CAPS. Lastly, some parts of the code: Kamailio.cfg: > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > debug=1 > log_stderror=no > rundir="/tmp" > memdbg=5 > memlog=5 > log_facility=LOG_LOCAL0 > # configure the prefix for all log messages > log_prefix_mode = 1 > log_prefix="{$mt $hdr(CSeq) $ci} " > /* number of SIP routing processes */ > children=32 # ------------- ALREADY TRIED WITH 8,16,24... no results..... > /* uncomment the next line to disable TCP (default on) */ > disable_tcp=no > /* uncomment the next line to disable the auto discovery of local aliases > * based on reverse DNS on IPs (default on) */ > # auto_aliases=no > /* add local domain aliases */ > # alias="sip.mydomain.com" > ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ ####### Modules Section ######## /* set paths to location of modules */ loadmodule "db_mysql.so" loadmodule "db_cluster.so" loadmodule "http_client.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "jansson.so" loadmodule "snmpstats.so" loadmodule "file_out.so" loadmodule "ctl.so" loadmodule "permissions.so" loadmodule "xhttp.so" loadmodule "xhttp_rpc.so" ####### Other Args and Env Vars ######### /* listen addresses */ include_file "listen.cfg" include_file "db_conn.cfg" # ----------------- setting module-specific parameters -------------- # --- DB Cluster params --- # minimum requirement is to have DBURL1 defined #!ifdef DBURL1 modparam("db_cluster", "connection" , DBURL1) #!endif #!ifdef DBURL2 modparam("db_cluster", "connection" , DBURL2) #!endif #!ifdef DBURL3 modparam("db_cluster", "connection" , DBURL3) #!endif #!ifdef DBURL4 modparam("db_cluster", "connection" , DBURL4) #!endif #!ifdef DBURL5 modparam("db_cluster", "connection" , DBURL5) #!endif modparam("db_cluster", "cluster", DBCLUSTER) # --- Permissions params --- modparam("permissions", "db_url", "cluster://k1") modparam("permissions", "db_mode", 1) #modparam("permissions", "reload_delta", 30) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ modparam("jsonrpcs", "fifo_name", "/tmp/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ modparam("jsonrpcs", "dgram_socket", "/tmp/kamailio_rpc.sock") modparam("jsonrpcs", "transport", 0) # ----- ctl params ----- /* set the path to RPC unix socket control file */ modparam("ctl", "binrpc", "unix:/tmp/kamailio_ctl") # ----- http_async_client params ----- modparam("http_client", "query_result", 0) modparam("http_client", "keep_connections", 0) modparam("http_client", "connection_timeout",2) modparam("http_client", "timeout_mode",2) modparam("http_client", "config_file", "/usr/local/etc/kamailio/http_client.cfg") # ---- SNMP Stats params ---- modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "sipEntityType", "redirectServer") modparam("snmpstats", "sipEntityType", "other") modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "MsgQueueMinorThreshold", 20) modparam("snmpstats", "MsgQueueMajorThreshold", 100) modparam("snmpstats", "dlg_minor_threshold", 20) modparam("snmpstats", "dlg_major_threshold", 100) modparam("snmpstats", "snmpCommunity", "kamailio") # ---- File_Out params ---- modparam("file_out", "base_folder", "/data/sabire002/log/kamailio/") modparam("file_out", "file", "name=stsh_requests;interval=1440;extension=.log") modparam("file_out", "file", "name=cdr;interval=1440;extension=.log") modparam("file_out", "file", "name=http;interval=1440;extension=.log") ####### Routing Logic ######## include_file "includes/reqinit.cfg" include_file "includes/handle_options.cfg" include_file "includes/handle_cancel.cfg" include_file "includes/handle_stir_shaken_stateless.cfg" include_file "includes/handle_http_rpc.cfg" request_route { $avp(START_TIME)=$utimef(%Y-%m-%d %H:%M:%S); $avp(GROUPID) = allow_address_group($si, $sp); if ( $avp(GROUPID) == 100 || !allow_address_group($si, $sp) ) { xlog("L_INFO", "INIT - $si:$sp is not in the allowed ACL Group ID !\n"); #sl_reply("401", "Address not allowed"); exit; }; if (is_method("ACK") ) { #&& t_check_trans() ){ exit; } if (is_method("INVITE")) { file_out("cdr","$rm|$ft|$tt|$ci|$rs|$rr|$Ts|$avp(START_TIME)|$fU|$fd|$si|$tU|$rU|$rd|$utimef(%Y-%m-%d %H:%M:%S)"); send_reply("100","Trying"); } route(HANDLE_OPTIONS); route(REQINIT); xlog("L_INFO"," ********** Route START ***********"); # log the basic info regarding this call xlog("L_INFO","=================================================== \n"); xlog("L_INFO","New SIP message $rm with call-ID $ci received $pr request $rm $ou source $si:$sp from $fu to $tu \n"); xlog("L_INFO","=================================================== \n"); route(HANDLE_STIRSHAKEN); route(HANDLE_CANCEL); if (method == "INVITE"){ route(RELAY); } } route[RELAY] { # Sends a 300 Multiple Choices back to the proxy that requested the routing lookup xlog("L_INFO","RELAY - send reply \n"); file_out("cdr","RELAY|$ft|$tt|$ci|$rs|$rr|$Ts|$avp(START_TIME)|$fU|$fd|$si|$tU|$rU|$rd|$utimef(%Y-%m-%d %H:%M:%S)"); send_reply("300", "Multiple Choices"); exit; } As for the HANDLE_STIRSHAKEN route, where the main process is: route[HANDLE_STIRSHAKEN] { if (!is_method("INVITE")) { return; } $xavp(requestTime) = $utimef(%s); $var(post) = $null; // resets and makes sure the $var(post) variable is null before usage [.... get some header values ...] $var(res) = http_connect("api1", "/stsh", "application/json", "$var(post)", "$var(result)"); jansson_xdecode($var(result), "json"); if ( $xavp(json=>sip-response-code) == 300 ) { remove_hf("Contact"); append_to_reply("Contact: $xavp(json=>contact)\r\n"); } else { # add error message to reply sl_reply("$xavp(json=>sip-response-code)","$xavp(json=>sip-response-text)"); exit; } } Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

8 73

How to use @via[idx] ?
by Benoit Panizzon 23 Dec '24

23 Dec '24

Hi https://www.kamailio.org/wikidocs/cookbooks/5.7.x/pseudovariables/#selname-… https://www.kamailio.org/wikidocs/cookbooks/5.7.x/selects/#viaihost I'm trying to list all via hosts: $var(dooh) = @via[1].host; xlog("L_INFO","******** VIA HOST 1: $var(dooh) \n"); This above works as expected I get the 2nd entry. $var(i) = 0; while ($var(i) < $hflc(Via)) { $var(dooh) = @via[$var(i)].host; xlog("L_INFO","******** VIA HOST: $var(dooh) \n"); $var(i) = $var(i) + 1; } This fails with syntax error. How do I specify the index of the via header I want to extract the host from? I also attempted $sel(via[$var(i)].host) or via[$var(i)] or $sel(hfl(via[$var(i)])).host but I found no variant not throwing an error. PS: Is there maybe another nice function to figure out if we have already seen (and how many times we have) this same message? 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 5

KAMI python3 and python3s not working / reloading
by trle.fuad＠gmail.com 20 Dec '24

20 Dec '24

Hello, I've started testing/learning KEMI and decide to use python3(s). I'm using default config with kamailio-basic-kemi-python3s.py (just deleted a couple of routes for simplicity). First thing I noticed is that reloading is not working. I've changed simple log line and did kamcmd app_python3s.reload. I'm compiling from source (inside Docker container, Debian bookworm as base), and have noticed it stopped working after thees commits for python3s: 62b4ee4a0d0b62b35c8bdf67e5daf9cbe9a28499 app_python3s: refactor GIL and thread state handling c81b2b3383c5900aba83672da024c812d8e6d89d app_python3s: initial support for free-threading Python And for python3 0ffe157bc13e7759ae1cee63a584fad4ac9eb38f app_python3: refactor GIL and thread state handling Am I missing some new parameter for module or compile flag? What I've noticed it's not only that it not reload, it also does not go through complete config file. While reloading, I see marking for reload. But nothing after that: 22(774) DEBUG: ctl [../../core/io_wait.h:368]: io_watch_add(): processing io_watch_add(0x748d1dfb83c0, 11, 3, 0x5f0adde454d0) - fd_no=1 22(774) DEBUG: ctl [io_listener.c:438]: handle_new_connect(): new connection (1) on /var/run/kamailio//kamailio_ctl 22(774) DEBUG: ctl [io_listener.c:499]: handle_stream_read(): bytes read: 29 22(774) INFO: app_python3s [apy3s_kemi.c:868]: app_python3s_rpc_reload(): marking for reload Python script file: /etc/kamailio/kamailio-basic-kemi-python3s.py (0 => 1) 22(774) DEBUG: ctl [io_listener.c:520]: handle_stream_read(): bytes processed: 29 22(774) DEBUG: ctl [io_listener.c:496]: handle_stream_read(): handle_stream read: eof on /var/run/kamailio//kamailio_ctl 22(774) DEBUG: ctl [../../core/io_wait.h:599]: io_watch_del(): DBG: io_watch_del (0x748d1dfb83c0, 11, -1, 0x10) fd_no=2 called

1 0

nethelper module - need clarification on how to handle uri alias on location contacts.
by Benoît Panizzon 20 Dec '24

20 Dec '24

Hi Gang When a CPE behind nat registers, I have the alias added to the AOR contact. It has format ip~port~transport Let's say: bob@1.2.3.4:5060;alias=5.6.7.8~7777~1 On an invite to that CPE behind nat, the contact is retrieved from the location record and I assumed I could call handle_ruri_alias in the branch route to that CPE and this would sent the call to that ip port via transport in the alias. I observe that the port is not taken into account. The INVITE to bob is sent to ip 5.6.7.8 but to port 5060 instead of 7777 In the example config, handle_ruri_alias() is only used for in dialog requests (has_totag). Have I misunderstood how CPE behind NAT should be registered and that registration resolved? -- 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 ______________________________________________________

1 0

Request for Feedback on an Installation Script for Kamailio (from Source)
by BROU STEPHANE FABIEN 19 Dec '24

19 Dec '24

Dear Kamailio Community, I hope this message finds you well. I have recently developed an install.sh script aimed at simplifying the process of installing Kamailio directly from source. My goal is to streamline the installation process while adhering to best practices for Kamailio deployment. I would greatly appreciate your feedback, suggestions, or any corrections to help improve the script’s robustness and ensure it aligns with the community’s standards. You can find the script in my GitHub repository: https://github.com/AL-CALL-LLC/KamailioBuilder.git Feel free to test it, share your thoughts, or open issues on GitHub for any recommendations or areas that need refinement. Thank you in advance for your time and valuable expertise. I am more than happy to provide further details or engage in discussions if needed. Best regards, Fabien Brou Developer

1 0

[print_dset(): no r-uri or branches] when working with async worker
by elhar.mohamed＠gmail.com 18 Dec '24

18 Dec '24

Hello, I'm getting this warning WARNING: <core> [core/dset.c:690]: print_dset(): no r-uri or branches when working with async worker. here is below the code used: request_route { if (is_method("INVITE")) { append_to_reply("Contact: <sip:+1234;myparam=test@$si:$sp>\r\n"); send_reply("302", "Moved Temporarily"); exit; } } Can you please help?

2 2

Multiple Branches and 3xx Status Codes
by Christian Berger 17 Dec '24

17 Dec '24

Hi, we have the following situation: Multiple UAs are subscribed to the proxy and get INVITEd via parallel forking. Now one of the UAs sends us a 302 status, we would like to handle this like a failure. So we'd like to CANCEL all the other branches, and forward the 302 status to the original INVITE. We have tried t_reply("302", "$T(reply_reason)") in the response, which almost solves this, however we cannot pass the required headers form the 302 response in it. BTW, could there have been a change in the default behavior of Kamailio? We think it used to "just work" in the past. Thank you very much 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 3

Development for v6.0.x is frozen
by Daniel-Constantin Mierla 17 Dec '24

17 Dec '24

Hello, the formal notification that the development for the next major version 6.0.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 6.0 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, new pull requests can still be done, but they will be merged after branching 6.0. Can still be done commits with documentation improvements, enhancements to related tools (e.g., kamctl, kamcmd), merging exiting pull requests at this moment, exporting missing KEMI functions and completing the functionality of the new modules added for 6.0. Once the branch 6.0 is created, new features can be pushed again to master branch as usual. From that moment, the v6.0.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 6.0 is going to be built at: * https://www.kamailio.org/wikidocs/features/new-in-6.0.x/ Upgrade guidelines will be collected at: * https://www.kamailio.org/wikidocs/install/upgrade/5.8.x-to-6.0.0/ Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.8.x to 6.0.x. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Freezing development for 6.0.x series
by Daniel-Constantin Mierla 16 Dec '24

16 Dec '24

Hello, I propose to aim freezing the development for 6.0.x series at the end of the 16th of December 2024 (Monday). New features that one wants to get in this release series have to be pushed to git repository or pull requests made for them. Afterwards usually follows a 4-6 weeks of testing till the first release 6.0.0. Unfreezing will happen earlier, after the first weeks of testing when the 6.0 branch will be created. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

ims_diameter_server: $diameter_response PV is null in the event_route[diameter:response]
by Grotti, Daniel 15 Dec '24

15 Dec '24

Hi list, I'm using kamailio 5.8 and playing with the ims_diameter_server module, by sending an async diameter request and trying to catch the reponse. Following the documentation, I'm sending the request with: route[SEND_ACR] { route(PREPARE_ACR); diameter_request_async("3", "271", "$var(json_acr)"); } route[PREPARE_ACR] { # here I build the AVP list in JSON $var(json_acr) = "[{"avpCode":277,"vendorId":0,"Flags":64,"int32":0},{...},{...}]"; } event_route[diameter:response] { xlog("Reply to diameter request is: '$diameter_response'\n"); } The ACR request is successfully sent and the the server is replying with ACA. The list of AVP are properly parsed, but when I enter the event_route to print the response, I get the $diameter_response = 'null': 2024-12-11T10:52:39.192951872+00:00 stderr F 11(19) DEBUG: cdp [worker.c:374]: worker_process(): worker_process(): [2] got task Q(2/2) 2024-12-11T10:52:39.193026280+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP((nil) < 0x7ff827633810 >0x7ff8276338c0);code=263,flags=40; 2024-12-11T10:52:39.193026280+00:00 stderr F DataType=1;VendorID=0;DataLen=52; 2024-12-11T10:52:39.193026280+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827633810 < 0x7ff8276338c0 >0x7ff827633970);code=268,flags=40; 2024-12-11T10:52:39.193026280+00:00 stderr F DataType=3;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193026280+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff8276338c0 < 0x7ff827633970 >0x7ff827633a20);code=264,flags=40; 2024-12-11T10:52:39.193026280+00:00 stderr F DataType=1;VendorID=0;DataLen=37; 2024-12-11T10:52:39.193026280+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827633970 < 0x7ff827633a20 >0x7ff8276340b0);code=296,flags=40; 2024-12-11T10:52:39.193026280+00:00 stderr F DataType=1;VendorID=0;DataLen=33; 2024-12-11T10:52:39.193124627+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827633a20 < 0x7ff8276340b0 >0x7ff827634160);code=259,flags=40; 2024-12-11T10:52:39.193124627+00:00 stderr F DataType=0;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193286132+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff8276340b0 < 0x7ff827634160 >0x7ff827634210);code=480,flags=40; 2024-12-11T10:52:39.193286132+00:00 stderr F DataType=0;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193286132+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827634160 < 0x7ff827634210 >0x7ff8276342c0);code=485,flags=40; 2024-12-11T10:52:39.193286132+00:00 stderr F DataType=0;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193286132+00:00 stderr F 11(19) DEBUG: ims_diameter_server [avp_helper.c:60]: avp2json(): AVP(0x7ff827634210 < 0x7ff8276342c0 >(nil));code=85,flags=40; 2024-12-11T10:52:39.193286132+00:00 stderr F DataType=0;VendorID=0;DataLen=4; 2024-12-11T10:52:39.193612877+00:00 stderr F 11(19) INFO: <script>: [event_route]: Reply to diameter request is: '<null>' Is there something I miss to set/configure? Thanks, Daniel Daniel Grotti Development Team Leader, Private 5G daniel.grotti(a)hpe.com<mailto:daniel.grotti@hpe.com> Hewlett Packard Enterprise HPE.com [cid:803b85f7-01fe-4c9c-a069-74d14a9ea122]

3 9

How to remove rr param on a reply?
by Benoît Panizzon 13 Dec '24

13 Dec '24

Hi Is there a way to remove an param from the top most rr header? I add a rr param towards the B CPE to store some information which is sent back on the rr header of the reply from B. So far this is what is intended and I also get this param in subsequent route headers from messages created by B and as the route header are remove towards A, this works fine. Unfortunately, in a case of a reply, the parameter is passed to A which is then sending back the param in the route header which would make it somehow complicated to figure out which side sent that parameter. I'm only interested if sent from B. So I wonder, is there a way to delete a rr param on replies to make sure I don't get it back in the ACK to the reply? -- 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 4

slot hongkong terbaik
by Womkevintanuwijaya＠gmail.com 13 Dec '24

13 Dec '24

<a href="https://slothongkong.pro/">slot hongkong</a> The best information from the world's largest number one blog!

1 0

registrar / usrloc / tm => 'load balanced' serial forking?
by Benoit Panizzon 13 Dec '24

13 Dec '24

Hi List With the dispatcher module, you can use different ways of load balanced and weighted serial or parallel forking. Is there a corresponding way to achieve load balanced serial forking to a destination set loaded from registered locations? Use case: Customer with a high availability set-up has two redundant PBX registered to the same AoR. Calls can not be sent in parallel as this would generate two concurrent calls to the end device but call should be distributed evenly between the two contacts registered with that AoR. 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 8

HTTP_CLIENT timeout
by Sergio Charrua 13 Dec '24

13 Dec '24

Hello all! For a STSH project, kamailio is making HTTP requests to a middleware (REST API) using HTTP_CLIENT module. I have set the parameter "timeout" to 2 second, but this increases PDD to too high levels, so I have set it to 1 second. Problem is that 1sec is also too high! The parameter only accepts integer values, so no way to set this to 0.5sec. I thought of having a way for kamailio to be aware that HTTP node #1 is down, so next request it will use HTTP node #2 instead. And once in a while, say, every 60 seconds, kamailio would check status of HTTP node #1 and, if OK, then it will switch back to HTTP node #1 and failover to node #2 when required. The HTTP_CLIENT, as far as I know, doesn't do that. So the question is: what would be the best approach to have this feature on Kamailio? i thought about using HTable and have a flag with value of the HTTP connection to use by default. At worst, only the 1st failover request would take 1 or 2 seconds to timeout....then the subsequent HTTP requests would use node #2. Any ideas? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

4 5

How to inspect/dump a stacked xavp? / How to stack two xavp on each other?
by Benoit Panizzon 10 Dec '24

10 Dec '24

Hi List I'm trying to inspect the content of the $xavp(tm_contacts) to figure out how t_load_contacts exactly behaves when called multiple times. $var(k) = 0; while ($xavp(tm_contacts[$var(k)]) != $null) { $var(out) = "nothing"; xavp_params_implode("tm_contacts[$var(k)]", "$var(out)"); xlog("L_INFO", "$cfg(route): tm_contacts idx: $var(k) => $var(out)\n"); $var(k) = $var(k) + 1; } My stack contains multiple entries. Unfortunately I struggle to access the lower entries. xavp_params_implode("tm_contacts[$var(k)]", "$var(out)"); does nothing when $var(k) > 0 What is the best way to dump a complete xavp stack? I suspect t_load_contacts does flush the complete stack. Is there a way to stack an xavp on top of another one? Something like: xavp_rm("contact_stack"); # Initialize Stack lookup("location", "sip:alice@example.com"); t_load_contacts(0); $xavp(contact_stack) = $xavp(tm_contacts); lookup("location", "sip:bob@example.com"); t_load_contacts(0); $xavp(contact_stack) = $xavp(tm_contacts); # contact_stack now contains alice + bob lookup("location", "sip:carlie@example.com"); t_load_contacts(0); # $xavp(tm_contacts) now contains only charlie. $xavp(tm_contacts) = $xavp(contact_stack); # $xavp(tm_contacts) now contains only alice, bob + charlie. t_next_contacts(); # Push contacts to destination list creating branches. t_relay(); # Invite all contacts 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 ______________________________________________________

6 10

segmentation fault when assigning xavp to another xavp (trying to stack them)
by Benoit Panizzon 10 Dec '24

10 Dec '24

Hi version: kamailio 5.7.6 (x86_64/linux) on Ubuntu from the: http://deb.kamailio.org/kamailio57 repository AOR alice has 4 contacts with q=0.5 AOR bob has 1 contact with q=0.5 I can reproduce a crash with this code snipplet: xavp_rm("contact_stack"); # Make sure XAVP does not exist. lookup("location", "sip:alice@example.com"); t_load_contacts(0); # after this, $xavp(tm_contacts) has four elements $xavp(contact_stack) = $xavp(tm_contacts); # Store the four elements on my stack # Verify, $xavp(contact_stack) now has four elements, copy successfull. lookup("location", "sip:bob@example.com"); # after this, $xavp(tm_contacts) has one element $xavp(tm_contacts) = $xavp(contact_stack); # Append the four previously saved contact with the one in $xavp(tm_contacts) if (!t_next_contacts()) { xlog("L_INFO", "$cfg(route): ERROR NO CONTACTS LOADED \n"); sl_send_reply("500", "Server Internal Error - Cannot get contacts"); } # Load contacts into destination set to parallel fork to five contacts. => segmentation fault! Shall I open an issue on github or am I performing a forbidden action? I'm doing some more test to find out where exactly the crash occurs. I'm not sure if it happens when adding to tm_contacts of when calling t_next_contacts. 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 1

uneven load distribution after gw goes down?
by David Villasmil 09 Dec '24

09 Dec '24

Hello guys, I'm seeing the dispatcher module used with ds_select_dst($avp(dsdst), "0") is distributing the calls to a failing gw, to the next available gw instead of distributing it. When a gw fails and has been marked as inactive, shouldn't it be taken out of the list and the list recomputed? is this not the case? Thanks all Regards, David Villasmil email: david.villasmil.work(a)gmail.com

2 2

Updated: Online devel meeting on matrix - 15:00 UTC, Dec 9, 2024
by Daniel-Constantin Mierla 09 Dec '24

09 Dec '24

Hello, we should consider an online devel meeting sometime soon to summarize what was done at (and still needs to be done after) devel meeting in Dusseldorf and plan a bit the targets for next major release 6.0. If considered useful, I propose Dec 9, 2024 (Monday) at 15:00UTC (16:00 Berlin/Paris/Madrid/Rome), but we can also look for other dates as well. Topics to be discussed can be added at: - https://github.com/kamailio/kamailio-wiki/blob/main/docs/devel/irc-meetings… Pull requests can be made by users without git access. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 2

WS and rtpengine_manage
by Knserbrave 09 Dec '24

09 Dec '24

Hi Kamailio Community, When testing rtpengine_manage and websockets, I noticed that if the rtpengine table in the database is not used, a new websocket is created for every call for a given dialog. That is, when an INVITE is received, an offer is sent to rtpengine. When the 200 OK or 18[3|0] is received, an answer is sent. However, the WS identifier often changes and as a result the request is not sent over the same websocket. This is problematic particularly when using FQDNs in the Kamailio configuration. When a load balancer is used between Kamailio and rtpengine, the answer may arrive at a different rtpengine in the cluster. After tracing through the source, I could not find a configuration parameter or set of triggers that I could manipulate to fix this. It leads me to believe it is only possible to reuse the websocket when the rtpengine table is used. Does anybody have more experience than I in this area that may be able to point me in the right direction, making load balancers for rtpengine_manage calls possible?

3 2

Potential Bug in Kamailio: Base64 Decode Transformation Not Working in Lua but Works in kamailio.cfg
by Dirk Brouwers | CM.com 28 Nov '24

28 Nov '24

Dear Kamailio Developers, I am writing to report a potential bug in Kamailio's handling of base64 decoding when using the Lua scripting interface (app_lua). It appears that base64 decoding behaves inconsistently between Lua scripts and kamailio.cfg. When using `$(var(encoded_json){s.decode.base64t})` in kamailio.cfg, where $var(encoded_json) contains the encoded base64 string: `eyJmb28iOiJiYXIifQ` it decodes to the correct `{"foo":"bar"}`. But when using the same transformation in Lua `KSR.xlog.xinfo("Decoded in Lua: ".. KSR.pv.get("$(var(encoded_json){s.decode.base64t})"))` it logs a corrupted encoded base64 string: `Decoded in Lua: ..]..#035z{b.?...#036..-....` Thank you in advance for looking into this issue. Please let me know if you need additional details or test cases. Kind regards, Dbrcm

3 4

"Intermittent Issue: Kamailio Presence Server Not Generating NOTIFY Packets"
by satyaprakash ch 27 Nov '24

27 Nov '24

Hi We have FreeSWITCH and Kamailio presence servers set up. FreeSWITCH sends XML RPC messages to the Kamailio presence servers for PUBLISH operations, and we receive 200 OK responses in return. These responses include the SIP-ETag and Expires headers. However, the Kamailio presence server is not consistently generating NOTIFY packets. This issue seems to occur randomly. Can you please suggest this for presence servers. Thank you in advance.

1 0

record_route() adds empty route header if enable_double_rr is 1
by Benoit Panizzon 26 Nov '24

26 Nov '24

Hi Gang How is record_route() determining if it needs to add two RR Header? Kamailio 5.8.4 I'm attempting to perform route header hiding by by rebuilding some of the topos functionality with perl. So towards the CPE I'm removing all existing RR header and want to replace them with one header containing an attribute which the CPE is then sending back and allows me to restore the Route header. But I came across an issue which an empty RR header being created with this (simplified) script: modparam("rr", "enable_double_rr", 1) perl_exec("hide_route"); # Stores Route-Set and sets $avp(rrattr) as id remove_hf("Record-Route"); msg_apply_changes(); loose_route(); # Required for record_route()? add_rr_param(";rr=$avp(rrattr)"); # Add Param to identify saved route set record_route(); I end up with: Record-Route: <sip:;r2=on;lr;rr=id007> Record-Route: <sip:157.161.x.x;r2=on;lr;rr=id007> The first header missing the IP address, the message being dropped by the CPE There is no transport change. If I set enable_double_rr to 0 only the 2nd RR header is created and this works as expected but I am not sure if some situations would need double RR. What causes the double RR to be created in my case? 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 1

app_perl add/remove HF?
by Benoît Panizzon 22 Nov '24

22 Nov '24

Hi List I'm looking into the perl module. Am I right, that I can not add or remove HF from an app_perl subroutine? I would need to pass an AVP from perl to tell Kamailio what to do? Goal is an attempt to re-build part of the topos functionality in perl but only handling RR and Route Header only on one leg, not on both. -- 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 ______________________________________________________

1 0

tcp_accept_haproxy option
by Wolter Eldering 22 Nov '24

22 Nov '24

Hi, I noticed that if I enable tcp_accept_haproxy kamailio still can accept connections without the PROXY-protocol header. The documentation on the wiki has the following note: "Please note that enabling this option will reject any inbound TCP connection that does not conform to the PROXY-protocol spec." https://www.kamailio.org/wiki/cookbooks/devel/core#tcp_accept_haproxy This could be true up until a specific version of kamalio but from 5.7.4+ this limitation is seems to be removed. Enabling tcp_accept_haproxy is breaking the outbound/rr module because it con’t find the socket based on the address and port from the flow token. The flow tokens created by a REGISTER on a connection without a PROXY header can be used. Flow tokens created from a connection with PROXY-protocol header will fail with "cannot find socket from flow-token” once loose_route is invoked. loose.c:561 /* First, force the local socket */ si = find_si(&rcv->dst_ip, rcv->dst_port, rcv->proto); if(si) set_force_socket(_m, si); else { LM_INFO("cannot find socket from flow-token\n"); return -1; } I’m not sure if the address and/or port values stored in the flow token are incorrect or the problem lies in the matiching in find_is(…). Wolter

1 0

RTC DevRoom at FOSDEM 2025
by Daniel-Constantin Mierla 21 Nov '24

21 Nov '24

Hello, FOSDEM'25 will host again a RTC DevRoom during the afternoon of February 1, 2025 (Saturday) -- more details and the call for presentation can be read at: - https://lists.fosdem.org/pipermail/fosdem/2024q4/003584.html I am not sure if I can participate (chances are more towards no, than to yes), but if anyone considers to go to the event, it will be good to submit a proposal to cover a bit Kamailio as well (it doesn't have to be entirely about Kamailio). Note that FOSDEM imposes a strict deadline this time for submissions (Dec 1, 2024), thus is no much time left. Should anyone in these groups plans to submit a proposal, let us know, so the others have an idea about it. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

TM-Module, ignoring
by Christian Berger 19 Nov '24

19 Nov '24

Hi, we would like to replicate a SIP-message ("REGISTER" in our case) to a set of nodes for further asynchronous processing. The message is already processed on the local Kamailio. We thought about using the tm-module in conjunction with the dispatcher module so we can reliably forward the message. This seems to work, however we get a new problem: Kamailio forwards the response it gets from those nodes. When we drop the response in the reply_route, it gets dropped before TM can see it. When we drop the response in the on_reply_route, it still will be forwarded. (adding or removing a header here works, however) $du is set to $null. Is there any clean way to suppress that response being forwarded back to the original requester? Best 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

2 2

Not able to use SRTP - from client to proxy
by Sajid Khan 18 Nov '24

18 Nov '24

Im using the module RTPengine. im having issues when ever i try to do a call using SRTP im using microsip i have checked the SDP everything ok. But when ever i place the call it gets rejected. if i disable SRTP then the calls work if (!rtpengine_offer("replace-origin replace-session-connection ICE=force RTP/SAVP RTP/AVP")) { I only need to encrypt client -> proxy from proxy -> asterisk should be plain rtp i hope someone can help me out thank you. The log in rtpengine shows this [1731055305.417896] ERR: [2a721ebf2c73414ab8e739d7bde912f9//1 port 14124]: [srtp] SRTP output wanted, but no crypto suite was negotiated

4 4

Kamailio 5.5.7 crash
by Mitesh Thakkar 18 Nov '24

18 Nov '24

Hi, I have recently migrated from Kamailio 5.2.5 to Kamailio 5.5.7. After a migration, the Kamailio crashed. Also, there were a lot of errors and warnings. Can someone help me to understand why this happened and what could be the solution? Logs (upside down): 18(397) INFO: <core> [main.c:859]: sig_usr(): signal 15 received 2(381) INFO: <core> [main.c:859]: sig_usr(): signal 15 received 30(409) INFO: <core> [main.c:859]: sig_usr(): signal 15 received 8(387) INFO: <core> [main.c:859]: sig_usr(): signal 15 received 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/tcp_read.c:1863]: handle_io(): io_watch_del failed for 0x7f526aa7f458 id 0 fd -1, state -1, flags 0, main fd 0, refcnt 0 ([]:0 -> []:0) 35(414) WARNING: <core> [core/tcp_read.c:1840]: handle_io(): F_TCPCONN connection marked as bad: 0x7f526aa7f458 id 0 fd -1 refcnt 0 ([]:0 -> []:0) 35(414) CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd -1, not in [0, 2) OS: Ubuntu 22.04.5 LTS \n \l # kamailio --version version: kamailio 5.5.7 (x86_64/linux) 272fdd 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: 272fdd compiled on 14:21:35 Oct 21 2024 with gcc 11.4.0 # dpkg -l |grep openssl ii openssl 3.0.2-0ubuntu1.18 amd64 Secure Sockets Layer toolkit - cryptographic utility Regards, Miteshkumar Thakkar

2 1

Pua PUBLISH method over xmlrpc/jsonrpc
by dries＠degendt.com 18 Nov '24

18 Nov '24

Hi all I've installed a (pretty old) presence_dfks module that allows setting the presence using the following command: kamctl fifo pua_publish sip:1000@10.10.99.254 3600 as-feature-event application/x-as-feature-event+xml . . . "<?xml version='1.0' encoding='ISO-8859-1'?><ForwardingEvent><device><notKnown/></device><forwardingType>forwardImmediate</forwardingType><forwardStatus>true</forwardStatus><forwardTo>1234</forwardTo></ForwardingEvent>" The "pua_mi" module was however removed in Kamailio and jsonrpcs/xmlrpcs are supposed to be an alternative. I've tried both the following calls, but neither does recognize the pua_publish/pua.publish as a valid method. Attempt with jsonrpc: curl -H "Content-Type: application/json" -X POST -d '{"jsonrpc": "2.0", "method": "pua.publish", "params": [""], "id":1}' https://sbctest.tel.redacted.xx:5061/RPC/ ERROR: jsonrpcs [jsonrpcs_mod.c:1422]: ki_jsonrpcs_dispatch(): method callback not found [pua.publish] Attempt with xmlrpc: curl -H "Content-Type: text/xml" -X POST -d '<?xml version="1.0" ?><methodCall><methodName>pua_publish</methodName><params><param><value><string>sip:jh@sbctest.tel.redacted.xx</string></value></param><param><value><string>7776000</string></value></param><param><value><string>as-feature-event</string></value></param><param><value><string>application/as-feature-event</string></value></param><param><value><string>.</string></value></param><param><value><string>a.1481534683.13958.6.7</string></value></param><param><value><string>sip:127.0.0.1:5080;transport=tcp</string></value></param><param><value><string>P-Flags: 0</string></value></param><param><value><string>Messages-Waiting: yesMessage-Account: sip:jh@sbctest.tel.redacted.xxVoice-Message: 2/0 (0/0)</string></value></param></params></methodCall>' https://sbctest.tel.redacted.xx:5061/RPC/ <?xml version="1.0"?> <methodResponse> <fault> <value> <struct> <member> <name>faultCode</name> <value><int>500</int></value> </member> <member> <name>faultString</name> <value><string>Method Not Found</string></value> </member> </struct> </value> </fault> Anyone who had some luck in this matter? Best regards, Dries

3 6

SIP NOTIFY over TLS fails
by dries＠degendt.com 15 Nov '24

15 Nov '24

Hi all I've installed a custom presence_dkfs module (https://github.com/tombeard/presence_dfks) in an effort to make our Grandstream devices (connected over TLS) display a server set callforward using Broadsoft's Device Feature Key Sync. The module was installed succesfully, but it appears that the NOTIFY upon SUBSCRIBE cannot be sent over TLS. This part seems to use the default presence module however so I'm taking my chances here to ask for some advice. I have already disabled verify_certificate in tls.cfg. The "dst addr: 193.19x.x.x:0" does seem to have an incorrect port as I should expect 5061? INFO: {1 600000 SUBSCRIBE 319937814-5062-3(a)BHC.DA.GB.CD} presence [notify.c:1744]: send_notify_request(): NOTIFY sip:544460@sbctest.tel.redacted.xx via sip:544460@172.30.61.23:5062;transport=tls on behalf of sip:544460@sbctest.tel.redacted.xx for event as-feature-event : 319937814-5062-3(a)BHC.DA.GB.CD ERROR: tls [tls_server.c:1312]: tls_h_read_f(): protocol level error ERROR: tls [tls_util.h:50]: tls_err_ret(): TLS connect:error:14094410:SSL routines:ssl3_read_bytes:sslv3 alert handshake failure (sni: unknown) ERROR: tls [tls_server.c:1316]: tls_h_read_f(): src addr: 172.30.61.23:5062 ERROR: tls [tls_server.c:1319]: tls_h_read_f(): dst addr: 193.19x.x.x:0 ERROR: <core> [core/tcp_read.c:1526]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f6d55b4b360 r: 0x7f6d55b4b488 (-1) Your advice is most appreciated! Cheers, Dries

4 7

dlq_req_within() usage for 3pcc-style reinvite media setup
by Alex Balashov 14 Nov '24

14 Nov '24

Hi, I am trying to use dlg_req_within() for a 3pcc-style call setup, to set up media via a reinvite: 1. Store original SDP offer from caller in $dlg_var()s; 2. Later, after e2e ACK is processed for initial INVITE transaction, send reinvite to callee using dlg_req_within(). I call rtpengine_offer() and feed it the original SDP offer (via read_sdp_pv modparam), and take the RTPEngine-transformed SDP and feed that to dlg_req_within(): dlg_req_within("callee", "INVITE", "application/sdp", "$var(sdp_from_rtpengine)"); 3. I have found that event_route[tm:local-response] does not allow me to capture the 200 OK / SDP answer to this reinvite; it is internally absorbed. However, it is possible to arm an onreply_route in event_route[tm:local-request], to receive the 200 OK: event_route[tm:local-request] { if(method == "INVITE" && has_body("application/sdp")) t_on_reply("REINVITE_REPLY"); } 4. My intent is for this reinvite reply handler to set off a reinvite to the caller side: onreply_route[REINVITE_REPLY] { if(method == "INVITE" && has_body("application/sdp") && t_check_status("200")) { ... $var(rtpengine_use_this_sdp) = $rb; rtpengine_answer("..."); dlg_req_within("caller", "INVITE", "application/sdp", "$var(sdp_from_rtpengine)"); } This ensures proper relay symmetry. Otherwise, the caller will continue to send RTP to the previous upstream endpoint of the call, prior to any reinvite. However, dlg_req_within() doesn't work in this later context, even though the documentation says it can be used from ANY_ROUTE. Kamailio doesn't complain, there is no error. It just doesn't initiate a reinvite to the caller. I considered the possibility that this may be because the calling scope is that of a pending reinvite transaction to the callee, but there is no obvious way to defer that into the future. If I send it to an async task worker, the transaction scope required for dlg_req_within() to know which dialog it's operating on will be lost. Any ideas appreciated, and thank you in advance! -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 2

KEMI / Python Issues
by Daniel W. Graham 14 Nov '24

14 Nov '24

Experimenting with KEMI for the first time, I ran into a few issues and hoping someone has feedback. kamailio 5.8.3 In the following python example, 'if ksr.is_method("ACK"):' should never be reached. Below is debug. if not ksr.siputils.has_totag(): ksr.info("has to tag") sys.exit() if ksr.is_method("ACK"): if ksr.tm.t_check_trans(): ksr.relay() sys.exit) else: sys.exit() ksr.sl.send_reply(404, "Not here") sys,exit() ..... DEBUG: app_python3s [apy3s_kemi.c:365]: sr_apy_kemi_exec_func_ex(): execution of method: siputils.has_totag DEBUG: siputils [checks.c:122]: has_totag(): no totag DEBUG: app_python3s [apy3s_kemi.c:368]: sr_apy_kemi_exec_func_ex(): execution of method: is_method DEBUG: app_python3s [apy3s_kemi.c:389]: sr_apy_kemi_exec_func_ex(): number of arguments: 1 DEBUG: app_python3s [apy3s_kemi.c:365]: sr_apy_kemi_exec_func_ex(): execution of method: sl.send_reply DEBUG: app_python3s [apy3s_kemi.c:389]: sr_apy_kemi_exec_func_ex(): number of arguments: 2 DEBUG: sl [sl.c:306]: send_reply(): reply in stateless mode (sl) ..... Second, here are a couple errors I have seen a few times but havent been able to identify the underlying issue, mainly hoping for more context on the meaning of 'execution of route type 1 with no name returned -1'. ERROR: app_python3s [apy3s_kemi.c:146]: apy3s_exec_func(): error exception occurred DEBUG: app_python3s [apy3s_kemi.c:230]: sr_kemi_config_engine_python(): execution of route type 1 with no name returned -1 Any input is appreciated. * dan

2 7

WG: [External] RE: Question regarding error message "failed to add new socket to the fd list"
by Floimair Florian 14 Nov '24

14 Nov '24

Thank you Henning! Here are the configured limits for Kamailio Limit Soft Limit Hard Limit Units Max cpu time unlimited unlimited seconds Max file size unlimited unlimited bytes Max data size unlimited unlimited bytes Max stack size 8388608 unlimited bytes Max core file size unlimited unlimited bytes Max resident set unlimited unlimited bytes Max processes 31543 31543 processes Max open files 16384 16384 files Max locked memory unlimited unlimited bytes Max address space unlimited unlimited bytes Max file locks unlimited unlimited locks Max pending signals 31543 31543 signals Max msgqueue size 819200 819200 bytes Max nice priority 0 0 Max realtime priority 0 0 Max realtime timeout unlimited unlimited us This is what I know so far: There seems to be a problem with siptrace module (going via TCP to a heplify-server on another host), but I’m not sure if this is a coincidence or the cause of the problems. I see quite a bit of these before the actual problem arises: Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322822]: ERROR: <core> [core/tcp_main.c:630]: _wbufq_add(): (591 bytes): write queue full or timeout (32498, total 43196, last write 0 s ago) Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322822]: ERROR: siptrace [../../core/forward.h:261]: msg_send_buffer(): tcp_send failed Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322822]: ERROR: siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: ERROR: <core> [core/tcp_main.c:4023]: handle_ser_child(): received CON_ERROR for 0x7fe054f8ae10 (id 727321), refcnt 2, flags 0x3096 At some point then all things go south and the log is flooded with these messages Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: CRITICAL: <core> [core/io_wait.h:596]: io_watch_del(): invalid fd 2244, not in [0, 482) Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: ERROR: <core> [core/tcp_main.c:4677]: handle_tcpconn_ev(): io_watch_del(3) failed: for 0x7fe054f8ae10, fd 2244 And then ultimately the above combined with the initially mentioned Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: CRITICAL: <core> [core/tcp_main.c:4264]: handle_ser_child(): failed to add new socket to the fd list And occasionally also something like this Nov 4 07:54:01 kamailio-prod-westeurope-2 /usr/sbin/kamailio[322829]: ERROR: <core> [core/io_wait.h:373]: io_watch_add(): trying to overwrite entry 2247 watched for 5 in the hash 0x5621dc404ee0 (fd:-601539712, type:22049, data:0x5621dc23c004) with (2247, 2, 0x7fe04f85c2a0) In total these add up to roughly 3600 lines per second (!!) in the log, so it is quickly flooded with these. Florian FLOIMAIR Software Development - Symphony Cloud Services (1568) Von: Henning Westerholt <hw(a)gilawa.com> Datum: Montag, 11. November 2024 um 19:52 An: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Floimair Florian <f.floimair(a)commend.com> Betreff: [External] RE: Question regarding error message "failed to add new socket to the fd list" CAUTION: This email originated from outside of the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. Hello Florian, Any other error messages before that error happens, e.g. something about the memory. Otherwise, it could be indeed a ulimit issue. First step would be to check the actual configured limits for the kamailio user. Cheers, Henning From: Floimair Florian via sr-users <sr-users(a)lists.kamailio.org> Sent: Montag, 11. November 2024 17:45 To: sr-users(a)lists.kamailio.org Cc: Floimair Florian <f.floimair(a)commend.com> Subject: [SR-Users] Question regarding error message "failed to add new socket to the fd list" Hi! We have recently had issues with one of our Production Kamailios. When those happened, the log was filled with the following message: CRITICAL: <core> [core/tcp_main.c:4528]: handle_new_connect(): failed to add new socket to the fd list Now I wonder what the best approach is to prevent this. We are using TCP/TLS only and I think this might be related to the file ulimit, but I am not sure about that. Shared memory is set to 512MB Can you give me a hint on what to look for? Thank you very much! P.S.: Sorry, I accidentally replied to a previous post of a different topic before which is totally unrelated (I think I should stop working for today 😉) FLORIAN FLOIMAIR Software Development - Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria [signature_2072127332] commend.com LG Salzburg / FN 178618z

2 1

Tell topos to create 'duplicate' route header from rr header with same ip
by Benoit Panizzon 14 Nov '24

14 Nov '24

Hi Gang I'm having another go with a dedicated kamailio running topos to replace a B2BUA SBC. I came across this situation: Server Topology: Kamaiolio SBC (topos) <=> Kamailio Registrar <=> Kamailio Router (DLG + CDRS) Core IP Address 10.0.0.1 Registrar IP 10.0.0.2 Two different CPE are registered on Registrar via SBC One CPE is calling the other CPE So the path the call is taking is: CPE A => SBC => Registrar => Core => Registrar => SBC => CPE B Invite A => B goes through, no issue. 180 Ringing, required 100rel B => A requires the route to be recorded therefore contains 3 record route statements: Record Route: <sip:10.0.0.2;lr> Record Route: <sip:10.0.0.1;lr> Record Route: <sip:10.0.0.2;lr> To ensure answer will get route back to the registrar => core => registrar and then the contact pointing to the SBC with topos to restore the last hop. Ringing B => A works as expected and is answered by A with PRACK. topos would now need set Route: in this PRACK message, according to the received Record Route Header. Unfortunately I fear Topos 'removes' the routes with duplicate host ip. I only get: Route: <sip:10.0.0.2> Route: <sip:10.0.0.1> Causing the Core not to be able to route the PRACK back to the registrar. To test, which line is being dropped, I added an mark to each one and determined the 3rd Record Router is being dropped by topos, probably considered a duplicate. I found no setting of the 'do not perform record route header dupe detection'. Interestingly, when I look at the database field topos_d.b_rr all three entries are present in the correct order. Has anyone come over the same issue and found a solution? Maybe related to: https://github.com/kamailio/kamailio/issues/3778 ? 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

Kamailio Developers Meeting, Nov 19-20, 2024, in Dusseldorf
by Daniel-Constantin Mierla 14 Nov '24

14 Nov '24

Hello, Kamailio SIP Server project is organizing another meeting of its developers and community members during November 19-20, 2024 (Tue-Wed), hosted again by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and contributors in order to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join, developer or user interested in helping the project. Please note we have a limited capacity of seats in the meeting room, the main policy for accepting participants being first come first server. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: * https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 4

Kamailio v5.8.4 Released
by Daniel-Constantin Mierla 12 Nov '24

12 Nov '24

Hello, Kamailio SIP Server v5.8.4 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.3. 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 v5.8.4. For more details about version 5.8.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/11/kamailio-v5-8-4-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 v5.8.4
by Daniel-Constantin Mierla 12 Nov '24

12 Nov '24

Hello, I am considering to release Kamailio v5.8.4 (out of branch 5.8) sometime next week, most likely on Tuesday, Nov 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

6 10

fail2ban complains about Kamailio syslog messages
by Juha Heinanen 12 Nov '24

12 Nov '24

When Kamailio message parser produces this kind of syslog message: 024-11-11T21:16:40.757364+02:00 box /usr/bin/sip-proxy[181398]: ERROR: <core> [core/parser/msg_parser.c:791]: parse_msg(): ERROR: parse_msg: message=<#026#003#001#006\340#001> fail2ban complains: Consider setting logencoding to appropriate encoding for this jail. Continuing to process line ignoring invalid characters: b'2024-11-11T21:06:15.347318+02:00 box /usr/bin/sip-proxy[181398]: ERROR: <core> [core/parser/msg_parser.c:791]: parse_msg(): ERROR: parse_msg: message=<#026#003#001#006\xc0#001>\n': 1 Time(s) Error decoding line from '/var/log/syslog' with 'UTF-8'.: 1 Time(s) Unable to retrieve Watch object associated to <_RawEvent cookie=0 mask=0x2 name='' wd=13 >: 3 Time(s) Would it be possible to somehow ensure that Kamailio syslog messages only contain valid UTF-8 characters? -- Juha

1 0

Last line of onreply_route (drops reply)
by James Browne 11 Nov '24

11 Nov '24

Hi all I noticed, in my config, that if the last line of the reply_route is a function that does not succeed, then the SIP response is not relayed and kamailio throws an error. I've searched documentation to indicate why this might be, but I can't find anything, so this looks to me like a bug. I use 5.8.1 for testing. Here's a sample config that shows what goes wrong. #################### #!KAMAILIO listen=udp:203.0.113.57:5060 debug=1 loadmodule "textops" request_route{exit;} reply_route{remove_hf_re("^X-Custom");} #################### Here's a sample SIP message that works. When I send this to udp:203.0.113.57:5060, kamailio relays it. #################### SIP/2.0 200 Testing Via: SIP/2.0/UDP 203.0.113.57:5060;branch=z9hG4bK-d8754z Via: SIP/2.0/UDP 198.18.0.157:5060;branch=z9hG4bK-d8754z T: <sip:c@s>;tag=676fbf45 F: <sip:c@s>;tag=676fbf45 X-Custom: lkjdfl i: testing-XXX CSeq: 1 INFO l: 0 #################### If I then remove the X-Custom header field from the SIP reply and try again, the message does not get relayed and I see this in the log. 5(1356) WARNING: <core> [core/receive.c:605]: receive_msg(): error while trying onreply script If I put a log command after the remove_hf_re(), then the message gets relayed. It looks like the reply_route always fails if the last function is not successful. Is this normal or is it a bug? James

3 5

Question regarding error message "failed to add new socket to the fd list"
by Floimair Florian 11 Nov '24

11 Nov '24

Hi! We have recently had issues with one of our Production Kamailios. When those happened, the log was filled with the following message: CRITICAL: <core> [core/tcp_main.c:4528]: handle_new_connect(): failed to add new socket to the fd list Now I wonder what the best approach is to prevent this. We are using TCP/TLS only and I think this might be related to the file ulimit, but I am not sure about that. Shared memory is set to 512MB Can you give me a hint on what to look for? Thank you very much! P.S.: Sorry, I accidentally replied to a previous post of a different topic before which is totally unrelated (I think I should stop working for today 😉) FLORIAN FLOIMAIR Software Development - Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria [signature_2072127332] commend.com LG Salzburg / FN 178618z

2 1

apt-key deprecation
by tom＠tomlynn.com 11 Nov '24

11 Nov '24

When I look at the debian repositories maintained by the project, there are instructions to import the public key of the repository for apt using this command: wget -O http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add - which results in: Warning: apt-key is deprecated. Manage keyring files in trusted.gpg.d instead (see apt-key(8)). gpg: no valid OpenPGP data found. Is there an alternate method that can be posted in the repository that will get better results? Thank you, TL

5 5

Unable to get body in multipart body
by Anthony Blandin 08 Nov '24

08 Nov '24

Hello, I want to get a body for content-type "application/isup" but I have the following error: 4(16) DEBUG: textops [textops.c:3394]: ki_get_body_part_helper(): found content type application/isup 4(16) ERROR: textops [textops.c:3397]: ki_get_body_part_helper(): no CRLF found after content type Here is the INVITE sent: INVITE sip:+883390000010022@[remote_ip]:[remote_port];user=phone SIP/2.0 From: <sip:+883390000010007@[local_ip];user=phone>;tag=[call_number] To: <sip:+883390000010022@[remote_ip];user=phone> Max-Forwards: 70 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Route: <sip:[local_ip]:[local_port];transport=udp;lr> Record-Route: <sip:[local_ip]:[local_port];transport=udp;lr> Call-ID: [call_id] CSeq: 1 INVITE P-Asserted-Identity: <sip:+883390000010007@[local_ip]:[local_port];user=phone> Accept: application/sdp,application/isup,multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK,UPDATE P-Charging-Vector: icid-value=IBCF-1-161460836711652742337;icid-generated-at=BC00.QX00.ORANGE-MSS.FR;orig-ioi=ORANGE-MSS.FR Supported: 100rel Content-Type: multipart/mixed;boundary=unique-boundary-1 MIME-Version: 1.0 Contact: <sip:[local_ip]:[local_port];transport=udp> Content-Length:[len] --unique-boundary-1 Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[local_ip_type] [local_ip] t=0 0 a=sendrecv m=audio [media_port] RTP/AVP 8 96 b=RR:0 b=RS:0 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=maxptime:40 --unique-boundary-1 Content-Type: application/isup;version=spirou;base=itu-t92+ Content-Disposition: signal;handling=required \x01\x11\x20\x01\x0a\x00\x02\x0c\x0a\x04\x10\x88\x33\x09\x00\x00\x01\x20\xf2\x0a\x0a\x84\x13\x88\x33\x09\x00\x00\x01\x00\x07\x1d\x03\x80\x90\xa3\x3f\x07\x83\x97\x26\x87\x41\x00\x00\x39\x04\x34\xc0\x3f\xc0\x00 --unique-boundary-1-- CRLF is well added at the end of Content-Type header, is anybody could help me to solve this issue? Regards, Anthony

1 0

[db_unixodbc] how to specify the database with unixodbc？
by 17603019528＠163.com 08 Nov '24

08 Nov '24

Hi Dear Community, I want to use db_unixodbc to connect to the dm8 database. Although I use Opensips, I found that this module db_unixodbc, Kamailio and Opensips are the same. My question is how to specify the database with kamailio？ I add "Database = sbc" into odbc.ini, But nothing happened。 ###### cat /usr/local/etc/odbc.ini [DM8] Description = ODBC DRIVER FOR DM8 Driver = DM8 ODBC DRIVER SERVER = 172.16.7.131 TCP_PORT = 5230 odbcinst -j unixODBC 2.3.12 DRIVERS............: /usr/local/etc/odbcinst.ini SYSTEM DATA SOURCES: /usr/local/etc/odbc.ini FILE DATA SOURCES..: /usr/local/etc/ODBCDataSources USER DATA SOURCES..: /root/.odbc.ini SQLULEN Size.......: 8 SQLLEN Size........: 8 SQLSETPOSIROW Size.: 8 ####### #################### modparam("usrloc", "db_url", "unixodbc://opensips:opensips#$@172.16.7.131:5230/dm8") #################### DBG:db_unixodbc:db_unixodbc_new_connection: opening connection: unixodbc://xxxx:xxxx@172.16.7.131/dm8 Nov 7 17:05:21 [145] DBG:db_unixodbc:db_unixodbc_new_connection: connection succeeded with reply <DSN=dm8;SERVER=172.16.7.131;UID=opensips;PWD=opensips#$;TCP_PORT=5230;> DBG:core:db_do_init: connection 0xfffd5bbbaf08 inserted in pool as 0xfffd5bbbb0e0 Nov 7 17:05:21 [145] ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= select table_version from version where table_name='dialog' Nov 7 17:05:21 [145] ERROR:db_unixodbc:db_unixodbc_extract_error: unixodbc:SQLExecDirect=42S02:1:-2106:Error in line: 1 ####################

2 1

Draining TCP/TLS SIP sessions from Kamailio and media sessions from RTPEngine
by Evgenii Buchnev 07 Nov '24

07 Nov '24

Hi Dear Community, Would appreciate your input into where to look further to explore my options and would be good to learn your experience with the following. There are many active-active kamailio nodes behind tcp load balancer. On the same VM with each Kamailio there is an instance of RTPEngine to manage Media/Transcoding/Etc for that Kamailio. This scales UP easily - new nodes just added behind tcp load balancer. To be able to scale down and/or update node configuration - I'd like to be able to drain sessions/calls properly and pass any active calls from current Kamailio and RTPEngine to other nodes. DMQ is not yet in place, although it's planned. What would be proper ways of migrating RTP sessions to other RTPEngines? Caveats? Also wonder how properly a tcp/tls based SIP session can be passed to another Kamailio? Thank you in advance for your input. Evgenii Buchnev

3 2

Roadmap to v6.0.0 - the next major release
by Daniel-Constantin Mierla 07 Nov '24

07 Nov '24

Hello, about 8 months since the last major release v5.8.x, therefore it is the time to plan a bit the next steps for getting v6.0.0 out. The Kamailio Developers Meeting is schedule in Dusseldorf, Germany during November 19-20, 2024, where we expect to push out a consistent number of updates. Then maybe we can wait for another 2 weeks or so to settle down/finish those updates. After that, if no other new features are planned by developers in short term, we can enter the testing phase, so the new major version can be released during the second part of January 2025 or first part of February 2025. Therefore, if you have new features under development that might require more than 1-1.5 months from now, and you want to get them in v6.0.0, let us know to plan properly. Other time lines suggestions to get to v6.0.0 are also welcome! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 1

Kamailio 5.5.7 alias not added to ACK
by Mitesh Thakkar 06 Nov '24

06 Nov '24

Hi, I was using kamailio 5.2.5. I have recently upgraded to kamailio 5.5.7. After upgrade, the alias is stop being add to the ACK of reinvite from upstream. Before upgrade, it was working fine. I’ll appreciate any help on this. Thank you. Regards, Miteshkumar Thakkar

1 0

SNMP setup
by Sergio Charrua 05 Nov '24

05 Nov '24

Hi all! I am having difficulties setting up SNMP, and mostly to figure out if what I have setup is correctly configured. This is what I have on Kamailio (5.8.2), which is set as stateless and as a redirect server (simply put, it receives INVITE from SBC, gets a route from routing services, modifies Contact header, and replies to SBC with 300 Multi Choice with new contact header): loadmodule "db_mysql.so" loadmodule "db_cluster.so" loadmodule "http_client.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "jansson.so" loadmodule "snmpstats.so" loadmodule "file_out.so" loadmodule "ctl.so" loadmodule "permissions.so" loadmodule "xhttp.so" loadmodule "xhttp_rpc.so" # ---- SNMP Stats params ---- modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "sipEntityType", "redirectServer") modparam("snmpstats", "sipEntityType", "other") modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "MsgQueueMinorThreshold", 1) modparam("snmpstats", "MsgQueueMajorThreshold", 1) modparam("snmpstats", "dlg_minor_threshold", 1) modparam("snmpstats", "dlg_major_threshold", 1) modparam("snmpstats", "snmpCommunity", "kamailio") I have set the *_minor_threshold values to bear minimum. Should I add anything else? The logs show the following messages: DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 60) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 64) returns address 0x7f03af3bcc50 frag. 0x7f03af3bcc10 (size=64) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 20) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 32) returns address 0x7f03af3bcd00 frag. 0x7f03af3bccc0 (size=32) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcc50), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(464) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bcc10 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcd00), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(471) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bccc0 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 60) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 64) returns address 0x7f03af3bcc50 frag. 0x7f03af3bcc10 (size=64) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 20) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 32) returns address 0x7f03af3bcd00 frag. 0x7f03af3bccc0 (size=32) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcc50), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(464) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bcc10 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcd00), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(471) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bccc0 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) INFO: snmpstats [utilities.c:116]: get_statistic(): failed to retrieve statistics for active_dialogs INFO: snmpstats [utilities.c:116]: get_statistic(): failed to retrieve statistics for active_dialogs The SNMP daemon is set as follows: - /etc/snmp]$ cat snmptrapd.conf # Example configuration file for snmptrapd # # No traps are handled by default, you must edit this file! # # authCommunity log,execute,net public # traphandle SNMPv2-MIB::coldStart /usr/bin/bin/my_great_script cold and on /etc/snmp/snmpd.conf: ############################################################################### # Further Information # # See the snmpd.conf manual page, and the output of "snmpd -H". agentaddress udp:161 rocommunity kamailio master agentx agentXSocket tcp:localhost:705 trap2sink 10.X.Y.Z netcool 162 When running tcpdump and listening on port 162, no traffic is captured....it seems to me that Kamailio is not sending anything to SNMP Any clue, anyone? What am I doing wrong? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 3

Function sdp_get gets extra blank line (when MIME is used)
by James Browne 05 Nov '24

05 Nov '24

The sdp_get() function from sdpops returns an extra blank line sometimes, which looks like a bug to me. When I have an SDP that's in MIME, there's an extra newline at the end of the extracted SDP. Here's a sample config. I use 5.8.1 for testing. #################### #!KAMAILIO listen=udp:203.0.113.57:5060 debug=1 loadmodule "sdpops" loadmodule "pv" loadmodule "xlog" request_route{exit;} reply_route { sdp_get("$avp(sdp)"); xlog("L_NOTICE", "[$ci] The SDP is [$avp(sdp)].\n"); } #################### Here are two sample SIP-message bodies. #################### v=0 o=PCD 1730 1730 IN IP4 198.18.114.148 s=PCD c=IN IP4 198.18.114.148 t=0 0 m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 #################### #################### --e61e7a6a-16a2-46c0-adbb-a41ba86d32a2 Content-Type: application/sdp Content-Length: 130 v=0 o=PCD 1730 1730 IN IP4 198.18.114.148 s=PCD c=IN IP4 198.18.114.148 t=0 0 m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --e61e7a6a-16a2-46c0-adbb-a41ba86d32a2-- #################### When the first body is received, the xlog correctly ends with this. m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ] with length 130 When the second body is received, the xlog incorrectly ends with this, which has an extra newline in it. m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ] with length 132 According to RFC2046, "The CRLF preceding the boundary delimiter line is conceptually attached to the boundary", but it looks like kamailio has a bug that incorrectly processes this. James

2 2

Simultaneous INVITE responses from multiple branches
by Anton Yabchinskiy 04 Nov '24

04 Nov '24

Hello, In our system we've encountered the following situation. There's a Kamailio proxy and a few REGISTER'ed devices behind the proxy. When there's an INVITE to these multiple devices, two of them simultaneously answer the call. Kamailio proxy accepts the first 200 OK from one of the devices and tries to CANCEL all other branches. But the second 200 OK device have already responded OK and doesn't acknowledge the CANCEL. So, there's now one established dialog with media, and another "established" one neither having media nor being correctly terminated. Looks like the similar scenario is described in section 3.1.2 of RFC 5407 (Example Call Flows of Race Conditions in the Session Initiation Protocol (SIP)). Is it possible to mitigate such scenarios in a Kamailio proxy? E.g., is it possible to send an additional BYE to these CANCEL'led branches? In current configuration, Kamailio proxy retrieves contacts with registrar:lookup() and does parallel forking with tm:t_relay().

2 2

Re: Dealing with flood of invalid SIP messages
by Juha Heinanen 04 Nov '24

04 Nov '24

I gave APIBAN a try. After a few days, no requests were banned by APIBAN, but during the same period, lots of request were banned by other checks in my config. That made me wonder if APIBAN really is effective. -- Juha

2 3

Error opening Kamailio's FIFO /run/kamailio/kamailio_rpc.fifo
by matilopi17＠gmail.com 01 Nov '24

01 Nov '24

Hello, This is kamailio version 5.8.2 and we`re having this issue every now and then. The error displayed is: ** ERROR: Error opening Kamailio's FIFO /run/kamailio/kamailio_rpc.fifo ** ERROR: Make sure you have loaded the jsonrpcs module and set FIFO transport parameters The problem is that the FIFO file /run/kamailio/kamailio_rpc.fifo doesn't exist, even though the "jsonrpc" module is in the Kamailio configuration, and the file is created when Kamailio service is restarted. But as mentioned, after few days the issue is repeated. I`ll appreciate any suggestion or advice of what I should check. Thank you!

4 5

Failed to parse the R-URI with topoh enabled
by Ale 29 Oct '24

29 Oct '24

Hi All, I think I have a problem with topoh, but I can't identify it. Kamailio 5.6.6, used as a stateless proxy, receives a 487 from a phone and propagates it correctly. Next comes the ack that should be forwarded back to the phone, but instead it doesn't forward it and produces the following error "pv_get_ruri(): failed to parse the R-URI" The error is generated by xlog("...$ru from $fu...\n"") as first line of request_route. If I disable topoh everything works perfectly.Topoh is only configured with mask_key only. Furthermore, not all phones generate this problem, at the moment there are some snoms. ACK sip:user@phone_ip:2048;line=kkqgarrj SIP/2.0 Via: SIP/2.0/UDP proxy_sip_ip;branch=z9hG4bK2ec.e7126a3134bef9d974f57dd43ebd4ea2.0 Route: <sip:kamailio_ip:5060;lr;received=sip:phone_ip:2048> Max-Forwards: 66 From: <sip:111111111@sip.example.com>;tag=pDc1BQ0B57Ujj To: <sip:user@sip.example.com>;tag=vslvowy4y2 Call-ID: 40a6c905-c0c8-4c20-b3f7-397b3fce58b6 CSeq: 90285584 ACK Content Length: 0 There is no difference between the ACK of the snom and other phones that work. The only difference I noticed is that the snom 487 contains Contact field in the header. Any suggestions? Thank you

3 5

How to handle e2e ACK with t_check_trans()
by Benoit Panizzon 28 Oct '24

28 Oct '24

Hi List To correctly route ACK belonging to a transaction (totag) the example code does: if ( is_method("ACK") ) { if ( t_check_trans() ) { route(RELAY); exit; } else { xlog("L_INFO", "$cfg(route): $rm NOT loose route NO transaction found DROP\n"); exit; } } Unfortunatels I fear that in a parallell forking scenario, the transaction is remove from memory as soon as the first 'negative' ACK is being processed so that the end 2 end ACK to the 200 OK of the branch that connected the call, never makes it through. I attempted to increase wt_timer to 10000 to have the transaction stay around 10 seconds but that does not solve the issue. I read that t_check_trans()'s result shall be considered more as a hint than as a confirmation of an existing transaction. Is there a clean / secure way to process e2e ACK and make sure they belong to a transaction? Or can I just safely remove the t_check_trans() call and route any ACK with a totag? 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 3

Dealing with flood of invalid SIP messages
by mayamatakeshi 27 Oct '24

27 Oct '24

Hi, I was going through some old company tickets that I am assigned to and found a case when possibly an attacker flooded our kamailio server with invalid sip messages like this: 2019-04-27T20:14:05.533554+09:00 IPX051 /usr/local/src/git/sip-router/kamailio[1732]: ERROR: <core> [parser/msg_parser.c:714]: ERROR: parse_msg: message=<[F#016sD#026Z<8D>97<F8><B5>;<A9><E7>-<D2>(<E2><F6> v;/#021k\<CC>8<B1>λ<F4>#004M<B6><BE><EC>#035#003<94><E1>=<A0><FF><E3><AF>Kwzr<8B>A#036B<D7>#027#023cu<82>Y<D4>#037<FB><AC>S_<C4>Qg<AB><DE>F<88>I#006<8C><FA><F4>~#y3G<C7>H<80>b<BC><AD>#035<89>#002<DB><C8>#001U<9E>#007<CB><F9>nT<E5><EE><8E><F1>#0144> At that time we manually banned the IP. But it would be helpful to have this done automatically by fail2ban. So I was thinking this log should include the src IP address. I looked at the latest kamailio commit and core/parser/msg_parser.c does this log the same way so I was thinking in opening an issue for this. But maybe this should be dealt with differently. Any ideas?

4 6

Issue with Retrieving Kamailio Statistics: Missing FIFO File and Errors
by satyaprakash ch 25 Oct '24

25 Oct '24

Hi, I'm encountering an issue while trying to retrieve Kamailio statistics on the server using the command kamctl stats. Instead of the expected output, I am receiving the following errors: kamctl stats: -e \E[37;31mERROR: Error opening Kamailio's FIFO /run/kamailio/kamailio_rpc.fifo -e \E[37;31mERROR: Make sure you have loaded the jsonrpcs module and set FIFO transport parameters Additionally, the file /run/kamailio/kamailio_rpc.fifo does not exist. Could you please provide suggestions ?

3 4

Handeling error returned by rtpengine_manage(...)
by Wolter Eldering 23 Oct '24

23 Oct '24

Hello, I have a question about the rtpengine module and how to use it via KEMI/python. Will processing of the native script stop when rtpengine_manage returns with an error? Or will it continue and will the SDP be forwarded unchanged? #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("SIP-source-address replace-origin replace-session-connection"); } else { rtpengine_manage("replace-origin replace-session-connection"); } #!else Regards, Wolter

3 4

Kamailio not deleteing rtp sessions
by Tihomir Culjaga 22 Oct '24

22 Oct '24

Hello, I'm using Kamailio + RTPEngine. Everything works well except one thing. I noticed rtp sessions are not being deleted after call termination. Instead, they remain lingering on RTPEngine waiting for a timeout. kamailio --version version: kamailio 5.5.2 (x86_64/linux) 55e232 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: 55e232 compiled on 14:13:04 Aug 25 2021 with gcc 8.4.1 rtpengine --version Version: 9.5.1.1-1.el8 rtpengine-ctl list totals tells me every session went to timeout :) --- snip --- Total statistics (does not include current running sessions): Uptime of rtpengine :914 seconds Total managed sessions :51 Total rejected sessions :0 Total timed-out sessions via TIMEOUT :51 Total timed-out sessions via SILENT_TIMEOUT :0 Total timed-out sessions via FINAL_TIMEOUT :0 Total timed-out sessions via OFFER_TIMEOUT :0 Total regular terminated sessions :0 Total forced terminated sessions :0 --- snip --- I'm using rtpengine_manage in two routes "route[RELAY]" and onreply_route[MANAGE_REPLY] . Do I need to add a specific rtpengine_delete somewhere ? route[RELAY] { ... $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (!has_totag()) { xlog("L_INFO", "NEXIOS: proto = [$proto], ru = [$ru]"); if ($proto =~ "ws" && !($ru =~ "transport=ws")) { xlog("L_INFO", "####################### FROM WS to any ########################\n"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal"; } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { xlog("L_INFO", "####################### FROM any to WS ########################\n"); setbflag(FLB_TOWEBRTC); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF direction=internal direction=external"; } else if (($proto =~ "udp") && $ru =~ "transport=udp" && $fs =~ "MY_IP2_ADDR") { xlog("L_INFO", "####################### FROM ?? to ?? $ru ########################\n"); } } # we engage RTPEngine if (has_body("application/sdp")) { xlog("L_INFO", "NEXIOS: # $xavp(r=>$T_branch_idx) #\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); } if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[MANAGE_REPLY] { ... $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (has_totag()) { xlog("L_INFO", "onreply_route proto = [$proto], ru = [$ru]"); if ($proto =~ "ws" && !($ru =~ "transport=ws")) { xlog("L_INFO", "onreply_route ####################### FROM WS to any ########################\n"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal"; } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { xlog("L_INFO", "onreply_route ####################### FROM any to WS ########################\n"); setbflag(FLB_TOWEBRTC); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF direction=internal direction=external"; } else if (($proto =~ "udp") && $ru =~ "transport=udp" && $fs =~ "MY_IP2_ADDR") { xlog("L_INFO", "onreply_route ####################### FROM ?? to ?? $ru ########################\n"); } } if (has_body("application/sdp")) { xlog("L_INFO", "onreply_route # $xavp(r=>$T_branch_idx) #\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); } } Any ideas are appreciated. Regards, Tihomir.

4 6

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