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42 participants
37519 discussions

failed to lock the memory pages (disable swap) WARNING
by Joel Serrano 27 Jul '20

27 Jul '20

Hi all, Can anyone give me some info on what this warning means: Jul 7 17:02:29 csbc02 csbc[16006]: WARNING: <core> [core/daemonize.c:596]: mem_lock_pages(): failed to lock the memory pages (disable swap): Cannot allocate memory [12] Yes, if I set mlock_pages=no I don't see the warning, but I'm trying to understand why can't it allocate with mlock_pages=yes. Joel.

4 4

Kamailio Installation Steps
by Abdul Rasheed 26 Jul '20

26 Jul '20

Hello All , I am trying to setup Kamailio on Debian . May i know which document should be followed for installing the latest version of Kamailio on Debian 10 Buster. ? Thanks and Regards Abdul Rasheed

2 2

Dialplan: using $avp("key") on subst rule with regexp capturing group
by João Vitor Arruda 24 Jul '20

24 Jul '20

Hello, I'm trying to use the dialplan with $avp("key") on match_exp and subst_exp but i'm getting an error message when there is a regex capturing group on subst_exp to be used on repl_exp This is the config used: modparam("dialplan", "match_dynamic", 1) $avp(s:PSTNAC) = "0"; Dialplan entry: dpid = 1 match_op = 1 match_exp = ^$avp(s:PSTNAC) match_len = 0 subst_exp = ^$avp(s:PSTNAC)(.*) repl_exp = 9\1 After executing dp_translate while the dialplan is properly matched the following error message occurs and the repl_exp didn't work as expected: ERROR: dialplan [dp_repl.c:422]: rule_translate(): illegal access to 1-th subexpr of subst expr (max 0) I've tried some other dialplan entries like: dpid = 1 match_op = 1 match_exp = ^$(avp(s:PSTNAC)[*]) match_len = 0 subst_exp = ^$(avp(s:PSTNAC)[*])(.*) repl_exp = 9\1 But that resulted in the same error message and no replacement either. Any suggestion on what I'm doing wrong here? Thanks, Joao Arruda

1 2

how to achive destination number based routing before falling back to dispatcher-module
by Christoph Russow 24 Jul '20

24 Jul '20

Hi all, im fairly new to Kamailio and currently setting up our new sip-server systems so, hello everyone! right now i have a Kamailio SBC which routes all calls using the dispatcher module to our internally setup application servers (freeswitch). i just got the idea that we might want specific routing based on destination phone number in addition to simple loadbalancing. can anyone point me in the right direction where i should look for something i can place before the dispatcher routing to route calls based on some entries in a database like destination number XYZ goes to server A, number ABC goes to server Z and so on. Or is there a way to select the dispatcher group id from the database based on destination number? best regards Christoph

4 9

HEP siptrace HEPv3 truncated and ignored by heplify-server (Homer 7)
by José Lopes 24 Jul '20

24 Jul '20

Hello all, I am using the siptrace module on Kamailio 5.3.5 and I am trying to monitor the SIP over websockets traffic with Homer 7.8.1 (heplify-server). I notice a strange behavior, some packets are not presented on Homer. I checked heplify-server and i notice the next warning: heplify-server: 2020/07/22 18:41:09.425513 decoder.go:103: WARN HEP packet length is 8192 but should be 11663 So I think the packets are truncated by 8192 (packet size), so they are ignored by heplify-server. I have done a tcpdump on the Homer machine and have the next output, where I can see some packets with size greater than 8192. 19:58:29.788927 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, bad length 11658 > 1472 19:58:29.793539 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 626 19:58:29.810733 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 579 19:58:29.811121 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 579 19:58:29.838824 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, bad length 11953 > 1472 19:58:29.852286 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 473 19:58:29.865388 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, bad length 3521 > 1472 19:58:29.875311 IP X.X.X.X.6666 > Y.Y.Y.Y.9060: UDP, length 562 Can you guide me to identify where the packets are truncated? Os melhores cumprimentos / Best regards, *José Lopes* Research and Development Technician Phone: +351 256 370 980 Email: jose.lopes(a)itcenter.com

2 2

new in upcoming Kamailio 5.4.0
by Henning Westerholt 23 Jul '20

23 Jul '20

Hello, I've updated the "New in version 5.4.0" document in our wiki. It is again an impressive list! * In this development period (started October 2019) almost 1500 commits have been added to the repository * 5 new modules have been added (pv_headers, kafka, secsipid, systemdops, dlgs) * Over 70 modules have been extended, many of them substantially (dispatcher, pv, tls, presence etc..) * Countless bugfixes and documentation extensions/fixes have been added as well * Many refactorings, extensions and architectural improvements were done in Kamailio core, tooling and modules You'll find the full details at the usual place: https://www.kamailio.org/wiki/features/new-in-5.4.x Many thanks to everybody that contributed to the Kamailio project. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

5 15

TAS recommendation
by BALL SUN 23 Jul '20

23 Jul '20

Hi We recently setup Kamailio as the IMS core in our test lab, and we would like to test scenarios with TAS, is there any recommendation? Does Kamailio able to setup as a SIP AS? if yes, any online instruction on how we can make it works? Thanks RBK

1 0

Dispatcher does not go to 2nd server
by Aristeidis Tsitras 22 Jul '20

22 Jul '20

I have a Kamailio and 2 asterisk servers. All users are created in both of the asterisk servers. I am forwarding the registration to asterisk. The problem is that it is always used on only one server from the list. Even if one goes to shutdown, then there is not any registration sent to the available server. Even if *some *of the extensions can be seen registered in both of the asterisk's, if the secondary goes down, then there are no services for the phones. I am attaching the kamailio.cfg. My dispatch list is: 1 sip:192.168.0.100:5080 0 0 maxload=20 2 sip:192.168.0.101:5080 0 0 maxload=20 In both of the asterisk servers i am using sip.conf to create users and s sip trunk for the Kamailio. Nothing special about it. I am looking to find what is wrong with my config and i cannot loadbalance/failover to the asterisk servers.

6 12

Siremis - PassThru Authentication
by Adam - UC-Reseller 21 Jul '20

21 Jul '20

I have recently installed Siremis And have set up a domain pointing to it, how do I configure it so registration requests to this domain are passed thru to the actual PBX Adam

2 1

P-Asserted-Identity in SIP INVITE from Kamailio IMS to PSTN (MGCF)
by BALL SUN 21 Jul '20

21 Jul '20

Hi We have a question. We are preparing a setup to bridge the SIP call from Kamailio IMS to PSTN, but we are puzzling how we can present the P-Asserted-Identity in the SIP INVITE. By default, the PAI is in sip:user@domain, but we believed we should use<tel> format in the PAI when bridge to PSTN. question is 1. should we have two PAI headers?, or 2. should we paste <tel> format in PAI User Part and domain in PAI Host Part?, or 3. Should we only use one PAI header with tel format? In our test setup, we are using FHoSS, and the IMPU sip:user@domain have below entry in "IMPUs from Implicit-Set" - sip:user@domain - tel:number Question is how we can extract the number from the HSS and presented in PAI? - RBK

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