Hello Brothers,
I've installed kamailio throw "apt install" on Debian and it's installed version: kamailio 5.6.3 (x86_64/linux).
Now I've a big problem that kamailio cannot running with TLSv1 and it has to be TLSv1.2+ as tls.cfg doc said:
# We do not enable anything else than TLSv1.2+
# over the public internet. Clients do not have
# to present client certificates by default.
How could I avoid this restriction please to enable TLSv1?
Thank you,
Hello,
Kamailio SIP Server v5.8.1 stable release is out.
This is a maintenance release of the latest stable branch, 5.8, that
includes fixes since the release of v5.8.0. 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 v5.8.1.
For more details about version 5.8.1 (including links and guidelines to
download the tarball or from GIT repository), visit:
 * https://www.kamailio.org/w/2024/04/kamailio-v5-8-1-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, April 18-19, 2024, Berlin -- kamailioworld.com
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
Hi
I am testing rtpengine recording with kamailio in lab. I have 1 endpoint
registered to Kamailio. My kamailio and rtpengine are running on same box.
I have another kamailio2 instance with 2nd endpoint to make a call to 1st
endpoint.
I have observed that Response dump for 'offer'/'answer' but further start
recording didn't invoke.
root@ip-kamailio:/etc/kamailio# rtpengine-ctl list sessions all
callid: J0WtgYIKd0Kwj1iL6BiFOg.. |
deletionmark: no | created: 1712120753 | proxy:127.0.0.1:52642 |
redis_keyspace:0 | foreign:no
callid: uH_ozKj18Nafshsm_sTElQ.. |
deletionmark: no | created: 1712120730 | proxy:127.0.0.1:52642 |
redis_keyspace:0 | foreign:no
callid: VYIw791dJska8PiIFspy8w.. |
deletionmark: no | created: 1712120716 | proxy:127.0.0.1:54804 |
redis_keyspace:0 | foreign:no
callid: nP89gWwjhsg_TSZrqu16nQ.. |
deletionmark: no | created: 1712120716 | proxy:127.0.0.1:56207 |
redis_keyspace:0 | foreign:no
root@ip-kamailio:/etc/kamailio#
I am new to kamailio and missing configuration. Suggest me the correct
configuration to invoke recording. attached config and logs
Regards,
Pawan
Hello guys,
I'm recently new player in Kamailio SIP server and want to use it as SIP
proxy server to Asterisk PBX server.
first of all, I've successfully registered my Asterisk to Kamailio as user
"test" to get its "location" because my Asterisk is behind NAT.
Simply now I want to relay SIP messages to this user's location to receive
them all in Asterisk. After I searched, I found this function rewritehost()
but it needs to be a host or IP address, but in my case I want some way or
function to relay to user "test" and then it well get the user location from
Kamailio datebase automatically?
I hope it's clear XD
I tried to find a way to do MSQL query in kamalio.cfg to get the user
contact from "location" table and then relay to the result but it did not
work. I guess that I did it in wrong way but finally the Kamailio tutorials
are so rare on google.
Thanks in advance,
Omar
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/
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>
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