sr-users July 2024

sr-users@lists.kamailio.org

42 participants
68 discussions

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

31 Jul '24

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

1 0

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

31 Jul '24

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

1 0

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

28 Jul '24

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

1 0

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

28 Jul '24

1 0

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

28 Jul '24

1 0

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

25 Jul '24

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

1 0

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

24 Jul '24

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

2 1

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

24 Jul '24

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

3 2

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

23 Jul '24

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

3 4

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

23 Jul '24

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

1 0

2026

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users July 2024