sr-users March 2024

sr-users@lists.kamailio.org

44 participants
42 discussions

by Joel Serrano

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

4 days, 21 hours

low performance with no apparent reason

by Sergio Charrua

Hi all! I have been doing some performance tests with Kamailio 5.7.4 and SIPp. The infrastructure is as follows:3 VMs running on VMWare ESXi running: UAC on 10.20.0.1 with SIPP-> Kamailio on 10.20.0.5 -> UAS on 10.20.0.3 The Kamailio VM has 6 dedicated vCPU of type Intel(R) Xeon(R) Silver 4216 CPU @ 2.10GHz and, 2 NICs and 4Gb RAM and MariaDB 10.6 as DB Backend., all running on a HP G380 host with a gazillion CPUs and a googol disk space! I currently have 3 scripts: - script #1 stateful with RTJson and simulating requests to routing engine and accounting - script #2 stateful but with just a simple routing to UAS, no rules, no DB, - script #3 stateless with a forward to UAS With script #3 I can go up to 2000CPS without issues with CPU at 37%! Above that value, I get retransmissions everywhere. On both scripts #1 and #2, the limit is 330CPS max after which I get a lot of retransmissions, while CPU/Core usage on Kamailio server stays below 10%. So I do not expect this to be a CPU issue. I could not understand why such (low) results, so I followed this article found at https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c <https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c> and created exact same scenarios, with kamailio script and SIPP templates available on the article, hoping for better results. But I get the same results: between 300 and 330CPS which is far, very far from the 7000CPS found in the article! I understand that I'm using VMs and probably the tests made for the article, which is pretty old already, were made on physical servers. Still, I would not expect 95% of lower performance! Any clue what could be the issue? I suspect NICs, but.... Any tips anyone could share? Thanks in advance! *Sérgio Charrua*

1 month, 2 weeks

Releasing v5.8.1

by Daniel-Constantin Mierla

Hello, with many countries having public holidays around Catholic Easter, I am considering to release Kamailio v5.8.1 (out of branch 5.8) on Wednesday, April 3, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. 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, April 18-19, 2024, Berlin -- kamailioworld.com

1 month, 2 weeks

Can we discuss the internals of Kamailio's routing logic and how to customize it for specific needs?

by Inextrix Technologies Pvt Ltd

This question digs into the heart of Kamailio's functionality. It asks about: Routing Logic: How Kamailio decides where to send SIP messages (calls, registrations, etc.) based on various factors like caller ID, destination number, or custom rules. Customization: How you can modify this built-in logic to fit your specific needs. This could involve using scripting languages, manipulating SIP messages, or defining custom routing rules. The discussion would be for developers who want to unlock Kamailio's full potential and tailor it for their unique VoIP deployments. Visit us for More Info : https://inextrix.com/services/kamailio-development/

1 month, 3 weeks

Callee details not visible in dialog

by egemen ulus

Hi, I am wondering why callee info is not visible in dialog details: Here is the case: client-1 > kamailio > two asterisks > kamailio > client-2 registration is on asterisk. dialog module is enabled and I want to see the asterisk IP that is determined by dispatcher module. I have tried to put dlg_manage function even after dispatch function, but did not help. I have also tried to update to header before dlg_manage function, it didn't help either. Any way to have asterisk IP in dialog details. It shows only this: { h_entry: 483 h_id: 8644 ref: 1 call-id: v489LdfOFD5mnlHupQHOvw.. from_uri: sip:1515@12.12.12.12;transport=UDP to_uri: sip:2222@12.12.12.12 state: 5 start_ts: 0 init_ts: xxxx end_ts: xxxx duration: xxxx timeout: 0 lifetime: 3600 dflags: 512 sflags: 0 iflags: 0 caller: { tag: d0283007 contact: sip:1515@1.1.1.1:64966;transport=UDP cseq: 1 route_set: socket: udp:172.22.10.10:5060 } callee: { tag: <null string> contact: <null string> cseq: <null string> route_set: <null string> socket: <null string> } profiles: { } variables: { } } Regards Get Outlook for Android<https://aka.ms/AAb9ysg>

1 month, 3 weeks

[Presence] Updating route information for SIP NOTIFY when re-SUBSCRIBE

by Denys Pozniak

Hello! In my architecture, SIP SUBSCRIBE messages can reach the Kamailio Presence server in several ways. And I noticed that re-SUBSCRIBEs messages do not update the record_route field in the active_watchers table (MySQL), so subsequent SIP NOTIFY messages do not inherit it and have the routes set of the initial SIP SUBSCRIBE message. Is there any way to change this behavior? modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "expires_offset", 0) modparam("presence", "max_expires", 1800) modparam("presence", "db_update_period", 30) modparam("presence", "clean_period", 180) modparam("presence", "send_fast_notify", 1) modparam("presence", "pres_htable_size", 32) modparam("presence", "subs_htable_size", 32) modparam("presence", "publ_cache", 0) modparam("presence", "notifier_processes", 0) # kamailio -v version: kamailio 5.1.2 (x86_64/linux) -- BR, Denys Pozniak

1 month, 3 weeks

RFC: move app_lua_sr to kamailio-archive

by Victor Seva

Hi, I would like to move app_lua_sr to kamailio-archive but I would like to hear if anyone out there has a solid reason not to do it. Since we have app_lua as KEMI for a long time I think is time to retire the old remains of app_lua Cheers -- ----------------------------------------------------------------- | ,''`. Victor Seva | | : :' : linuxmaniac(a)torreviejawireless.org | | `. `' PGP: 8F19 CADC D42A 42D4 5563 730C 51A0 9B18 CF5A 5068 | | `- Debian Developer | -----------------------------------------------------------------

1 month, 3 weeks

Dialog and DMQ on restart

by Björn Klasen

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

1 month, 3 weeks

RTP Engine with Kamailio

by Vishal Pai

Hello Everyone. I am looking for documentation to have multiple RTP Engine servers connected with kamailio using db url and balance the load accordingly. Thanks

1 month, 4 weeks

Kamailio to aid in seamless data center migration?

by Brad

I need to move a few dozen FreePBXen with some commercial modules running in individual VMs to a new data center. I’m trying to work out a plan to move the PBXes to the new data center in a way that will be transparent to the endpoints, or at the very least with the absolute minimum of downtime. Some of the installations are rather old, and there’s a handful of peculiarities on each, so the typical FreePBX backup/restore process hasn’t gone smoothly, at least in our tests. My current train of thought is to put a clone the VMs in the new data center and use Kamailio to route the SIP traffic to the new servers/IPs. I’ve never used it before, so I may be barking up the wrong tree, but it /appears/ to do what I’m suggesting. If so, I’m thinking I can install Kamailio on each VM, point it to the local asterisk/FreePBX initially, and clone the VM. Then, after the new instance is up and running, point Kamailio on the original VM to the cloned VM's asterisk, after which I can make appropriate DNS changes. Another option would be to stand up a single Kamailio server and redirect SIP traffic destined for individual asterisk servers to it at the router. Endpoints are mostly Yealink, and I’m not sure if they’ll feel the need to restart when the registration/SIP server’s IP changes, but a quick bounce when not in use isn’t the end of the world. I’d very much like to avoid having to send an update to each phone manually, but I can script a SIP NOTIFY if required. Anything stupid, wrong, ignorant, or just smelly about this tactic? Or, for that matter, any other suggestions?

1 month, 4 weeks

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sr-users March 2024