Hello!
Is it possible to find out the name of the onreply_route that was set
before?
Something like this:
t_on_reply("MANAGE_REPLY");
...
if ( t_is_set("onreply_route") ) {
get_onreply_route_name();
...
}
--
BR,
Denys Pozniak
Hello,
I encountered a problem stopping Kamailio with FIPS OpenSSL:
Program terminated with signal SIGSEGV, Segmentation fault.
#0 0x00007ff7292380ac in OPENSSL_sk_pop () from /lib64/libcrypto.so.3
Missing separate debuginfos, use: dnf debuginfo-install
kamailio-5.7.3-4816.x86_64
(gdb) bt
#0 0x00007ff7292380ac in OPENSSL_sk_pop () from /lib64/libcrypto.so.3
#1 0x00007ff72914bf5b in conf_modules_finish_int () from /lib64/libcrypto.so.3
#2 0x00007ff72914c694 in CONF_modules_unload () from /lib64/libcrypto.so.3
#3 0x00007ff7291efff9 in OPENSSL_cleanup () from /lib64/libcrypto.so.3
#4 0x00007ff72954702b in ?? ()
#5 0x0000000100061c08 in ?? ()
#6 0x00007ff7190566c8 in ?? ()
#7 0x00007ffccf196a20 in ?? ()
#8 0x000000000071da8a in futex_release (lock=0x7ff729f08b50 <syslog>)
at core/mem/../mem/../futexlock.h:134
#9 0x00000000006e9448 in destroy_tls () at core/tls_hooks.c:75
#10 0x000000000041f278 in cleanup (show_status=1) at main.c:594
#11 0x0000000000420af1 in shutdown_children (sig=15, show_status=1) at
main.c:721
#12 0x0000000000421717 in handle_sigs () at main.c:752
#13 0x0000000000430c88 in main_loop () at main.c:1988
#14 0x0000000000439d13 in main (argc=14, argv=0x7ffccf1973f8) at main.c:3212
(gdb)
Environment:
Oracle Linux Server 9.3
Kamailio 5.7.3
yum list --installed | grep ssl
openssl.x86_64 10:3.0.7-24.0.3.el9_fips
@tools
openssl-libs.x86_64 10:3.0.7-24.0.3.el9_fips
@tools
openssl-pkcs11.x86_64 0.4.11-7.el9
@anaconda
xmlsec1-openssl.x86_64 1.2.29-9.el9
@AppStream
What can I do for further investigation?
Thanks
Hello,
Kamailio SIP Server project is organizing another meeting of its
developers and community members during November 19-20, 2024 (Tue-Wed),
hosted again by sipgate.de in Dusseldorf, Germany.
The event is intended to facilitate the interaction between Kamailio
developers and contributors in order to offer a convenient environment
for working together on several topics of high interest for the project,
including writing code for Kamailio and its tools, improving
documentation, or discuss about future development.
Everyone from the community is welcome to join, developer or user
interested in helping the project. Please note we have a limited
capacity of seats in the meeting room, the main policy for accepting
participants being first come first server. Also, very important to be
aware that this is not an event to learn how to use Kamailio.
More details about the event, the venue, how to register, are available at:
* https://www.kamailio.org/w/developers-meeting/
Looking forward to those two intensive hacking Kamailio days in Dusseldorf!
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Hello!
I need to disable topos for one specific SIP trunk (in-out), it looks like
it’s enough to use event_route with IP address filtering.
But for some reason, the incoming INVITE from the peer still gets processed
by topos and I also don’t see a mention of [msg-incoming] in the logs, only
this:
WARNING: <script>: [msg-outgoing] OPTIONS/you/1.1.1.1
Code snippet:
loadmodule "topos.so"
modparam("topos", "db_url", DBURL)
modparam("topos", "contact_mode", 1)
modparam("topos", "header_mode", 1)
modparam("topos", "methods_noinitial", "OPTIONS,SUBSCRIBE,PUBLISH")
modparam("topos", "dialog_expire", 7210)
modparam("topos", "rr_update", 1)
modparam("topos", "event_mode", 5)
/*
1 - execute event_route[topos:msg-outgoing]
2 - execute event_route[topos:msg-sending]
4 - execute event_route[topos:msg-incoming]
8 - execute event_route[topos:msg-receiving]
*/
request_route {
....
event_route[topos:msg-outgoing] {
if ( $sndto(ip) == "1.1.1.1" ) {
xlog("L_WARN","[msg-outgoing] $rm/$rU/$sndto(ip) \n");
drop;
}
}
}
event_route[topos:msg-incoming] {
if ( $si == "1.1.1.1" ) {
xlog("L_WARN","[msg-incoming] $rm/$rU/$si \n");
drop;
}
}
# kamailio -v
version: kamailio 5.7.1 (x86_64/linux) 1cf389-dirty
--
BR,
Denys Pozniak
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)
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,
I have integrated Microsoft teams and freepbx using Kamailio version 5.8 as
my SBC.
Calls from freepbx to teams are working fine with audio and I am having
challenges with calls from teams to freepbx. The calls rings and after
answering there is only way audio and the call drops off after thirty
seconds. My pbx is behind a microtik router and I have port forwarded my sip
port and rtp ports to freepbx and have also disabled sip alg. My Kamailio is
hosted in AWS. I have attached both my pcap from Kamailio, freepbx and my
Kamailio.cfg. I am thinking It's a misconfiguration challenge or something
else. Please help I am failing to see my challenge.
Best Regards,
Hello Kamailio Community,
I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on:
-Optimizing Kamailio’s configuration for high availability and performance.
-Efficiently managing load balancing and failover.
-Recommended hardware specifications for large-scale operations.
-Common pitfalls to avoid when scaling up.
Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced.
Thank you!
Maria
[aws](https://www.igmguru.com/cloud-computing/aws-developer-certification-tr…
I have integrated Microsoft teams and freepbx using Kamailio version 5.8 as
my SBC.
Calls from freepbx to teams are working fine with audio and I am having
challenges with calls from teams to freepbx. The calls rings and after
answering there is only way audio and the call drops off after thirty
seconds. My pbx is behind a microtik router and I have port forwarded my sip
port and rtp ports to freepbx and have also disabled sip alg. My Kamailio is
hosted in AWS. Below are google drive links logs for the incoming call in
Kamailio , freepbx and my Kamailio.cfg. I am thinking It's a
misconfiguration challenge or something else. Please help I am failing to
see my challenge.
kamailio.pcap -
https://drive.google.com/file/d/1aoJhksgwcH-52unwxuyzn8GAmG9PEnOM/view?usp=s
haring
freepbx.pcap -
https://drive.google.com/file/d/1JJXs9UOUFak5Y4uWI_ARB56vQfVYfSDj/view?usp=s
haring
kamailio.cfg -
https://drive.google.com/file/d/1v2ZtylOf8fIMYH4OckGmnP489PO7WI7J/view?usp=s
haring
Best Regards