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37509 discussions

Prioritize specific codecs
by Temilolu Fayomi 17 Feb '25

17 Feb '25

Hello, I am working with inbound calls on kamailio (specifically fax), and was wondering if there was a way to re-prioritize the codecs found in the sdp of an invite. I've looked at using the sdpops module, but haven't been able to re-arrange the priority of the codecs listed. I've tried removing codecs to help incoming faxes switch over to t38 per my upstream providers instructions (calls will do a t38 re-invite if the g729a is the highest priority). I am just looking for which direction to go from here or if I misunderstand the nature of faxing and kamailio. Thank you in advance, Temilolu Fayomi

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Constant crashes across versions
by Jon Bonilla (Manwe) 17 Feb '25

17 Feb '25

Hi all I have a setup of several proxies behind a load balancer and all of them have several coredumps every day. I've tried versions 5.3, 5.5 and 5.8 and all of them crash I didn't have dbgsym packages installed but now I've built a 5.8.4 version in one of them and can gdb the coredumps. Can't see anything particular and I'm wondering if I'm doing anything wrong. I'm descompressing the coredump with lz4cat. Sometimes I see that the system generates 3 coredumps at the same time. I've tried running gdb in one of them and exec "bt full" I usually see "can't access memory" in some of them but inb others I don't see anything relevant. I'm attaching one coredump. I don't even know if I'm doing it properly. Could you please guide me to how to debug what's going on? thanks

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Info: sipexer v1.2.0 released
by Daniel-Constantin Mierla 11 Feb '25

11 Feb '25

Hello, a short note to announce that that sipexer v1.2.0 was released: - https://github.com/miconda/sipexer/releases/tag/v1.2.0 It is a modern and flexible SIP testing cli tool that can be used to send SIP signalling messages over UDP/TCP/TLS/WebSocket. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, March 24-28, 2025 -- asipto.com Kamailio World Conference, May 12-13, 2025, Berlin -- kamailioworld.com

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DNS Unlinked Warnings
by John Larsen 10 Feb '25

10 Feb '25

Hi Everyone, We recently attempted an upgrade from kamailio 5.5.7 to 5.7.6. We immediately started to receive a log of warning logs similar to `dns_hash_put_shm_unsafe(): unlinked item 0x7f08799dcd90` The code shows that this warning log was added to the DNS cache in september 2023 - but its unclear what it means to have an unlinked item exactly? Is anyone able to shed any light on how DNS entries might become unlinked, and what the implications of it are? thanks John Larsen John Larsen Tech Lead EJohn.Larsen(a)symbio.global W symbio.global This communication is intended only for the person to whom it is addressed and may contain confidential material. If you received this communication in error, please inform the sender immediately and delete all copies. Please think of the environment before printing this email.

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switch between "sips:" and "sip:" schema
by James Lipski 10 Feb '25

10 Feb '25

Hello, I have a kamailio installation where it's basically doing transport protocol translation. To go into more detail, I have a UA communicating over TLS towards kamailio/ and kamailio forwards the request to a freeswitch installation via UDP, so call-flow is as follows: UA <-- (TLS) --> Kamailio <-- (UDP) --> Freeswitch For the most part, this does work/ it's only when I use a UA that uses "sips:" as the schema instead of "sip:" where issues occur -- in my current topology, freeswitch has some weird behavior where it doesn't disconnect the call properly. At least based on my testing, I believe the issue is relating to the schema -- as the UA in question does have the ability to switch between schemas/ when using "sip:" instead of "sips:" over TLS, no issues with the same call-flow. In the event that I do come across a UA that doesn't have that ability, I was wondering if kamailio has the ability to switch schemas when it sends traffic over the UDP connection. I'm able to replace most aspects of the packet with "sip:" however the only header which contains the "sips:" schema is the contact header. With this alone, it propagates to the B-leg side (UDP side) of the call; and will cause freeswitch to not disconnect calls properly. I'm using the topos module to obfuscate my topology, not sure if that matters. -------- For example, on the A-leg side of the call, kamailio receives this: INVITE sips:12125551000@dev-sip.server.com:5061 SIP/2.0 Via: SIP/2.0/TLS 172.16.1.194:5061;branch=z9hG4bK236557042;rport;alias From: <sips:100000000@dev-sip.server.com>;tag=444048970 To: <sips:12125551000@dev-sip.server.com> Call-ID: 1326419916-5061-4(a)BHC.BG.B.BJE CSeq: 21 INVITE Contact: <sips:100000000@172.16.1.194:5061;transport=tls> Proxy-Authorization: Digest username="100000000", realm="dev-sip.server.com", nonce="Z6Ig0WeiH6Vdxtvej0I3XGCTUAEfh+0/", [uri="sips:12125551000@dev-sip.server.com](mailto:uri=)", response="43c1487d077db36a70fffdd1e25b3b52", algorithm=MD5 Max-Forwards: 70 User-Agent: Grandstream HT812 1.0.57.1 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 1034 v=0 o=100000000 8000 8000 IN IP4 172.16.1.194 s=SIP Call c=IN IP4 172.16.1.194 t=0 0 m=audio 5004 RTP/SAVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=crypto:1 AEAD_AES_256_GCM inline:UlX3PAjNBP/aQwtwyf5ymsfJciB+VdFxMElO5zLVooUqmcEzZ8YyQQk9stI=|2^32 ------------------------------------------------------ On the B-leg side of the call, kamailio forwards the following to freeswitch: INVITE sip:12125551000@fs.server.com SIP/2.0 Via: SIP/2.0/UDP 10.64.52.113:5080;branch=z9hG4bK1552.89ec9666294034a5611f908f2a6a427e.0;i=f From: "Test" <sip:16461234567@dev-sip.server.com>;tag=444048970 To: <sip:12125551000@fs.server.com> Call-ID: 1326419916-5061-4(a)BHC.BG.B.BJE CSeq: 21 INVITE Max-Forwards: 15 Supported: replaces, path, timer, eventlist Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 316 Contact: <sips:btpsh-67a21ca2-2dbe86-1@10.64.52.113:5080> v=0 o=100000000 8000 8000 IN IP4 10.64.52.113 s=SIP Call c=IN IP4 10.64.52.113 t=0 0 m=audio 13060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=ptime:20 -------------------------------------------------- As with the above, I was wondering if there is a way within kamailio to switch schemas on the b-leg side of this call example; specifically is there a way to update the contact header so it doesn't use the "sips:" schema when it traverses the UDP connection. Thank you for you time.

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Auth_ephemeral module setup
by Anoop L 04 Feb '25

04 Feb '25

Hi, I'm trying to understand and setup Auth_ephemeral for authentication. My goal is to authenticate users dynamically using an web service for that I bumped into auth_ephemeral module. In the module how is a request made to the web service? Are username and password fetched from Pseudo-Variables variable? Is this module to be worked with xhttp module for http requests? Any help is much appreciated.

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Auth_ephemeral Module Setup
by anoop96115＠outlook.com 04 Feb '25

04 Feb '25

1 0

Testing Kamailio IMS Modules with IPSec as P-CSCF
by Saeed Mogharian 03 Feb '25

03 Feb '25

Hi, I’m looking for suggestions on how to test Kamailio IMS modules, specifically for the IMS registration process over IPSec when Kamailio is configured as the P-CSCF node. Are there any scripts or IMS clients available that can help with this? Thanks in advance!

2 1

Kamailio 6.0 presentation at FOSDEM 2025
by Henning Westerholt 31 Jan '25

31 Jan '25

Hello, just a quick announcement, that tomorrow afternoon will be a presentation about Kamailio 6.0 at the FOSDEM conference in Brussels, Belgium [1]. Usually all the talks in this room are streamed online as well. I am happy also to meet during the conference to chat about Kamailio or other topics, just send me a (direct) message. Best regards, Henning [1] https://fosdem.org/2025/schedule/event/fosdem-2025-5978-kamailio-6-0-develo… -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

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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