Hello,
Just to confirm, as I quickly checked into a code of 5.8.x branch, there
were no modification on app_python3 module to comply with tls_threads_mode
mechanism?
Asking this beforehand posting a bug, that if using a quite simple script
into module, with tls_threads_mode = 2 Kamailio just hangs at the start and
become unresponsive.
And by simple I mean something like this
class PyKamailio:
def __init__(self):
self.test = 1
def child_init(self, rank):
"""Executed by app_python module when a new worker process (child)
is initialized by Kamailio."""
return 1
def mod_init():
""" Executed by app_python module when it is initialized by Kamailio."""
return PyKamailio()
with tls_threads_mode = 1 and 3 it loads, but if add some HTTPS calls
inside the script (even not the calls, but importing the methods, that are
making calls with import), it just start failing on TLS load, despite that
on packets processed (OPTIONS) no python_exec() is called, just from the
fact of having HTTPS calls into Python script.
--
Best regards,
Ihor (Igor)
Hello Everyone
I am using kamailio with RTP engine for wss to udp to connect my asterisk
server. It all works well like wss to UDP registration via kamailio to
asterisk. Here I am facing an issue with call hold i.e if I hold the call
from B leg it gets disconnected.
Any advice why it get failed and hangups
Thanks
Hello,,
I am using this docker file for installaing kamailio. I wana use it with open5gs for trynig VoNR , I used before it for VoLTE it works but right now I have this error in installation process.
icscf | 0(34) ERROR: <core> [core/sr_module.c:527]: ksr_locate_module(): could not find module <maxfwd> in </usr/lib64/kamailio/modules_k/:/usr/ lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux-gnu/kamailio/modules/:/usr/local/lib64/kamailio/ modules>
icscf | 0(34) CRITICAL: <core> [core/cfg.y:4008]: yyerror_at(): parse error in config file /etc/kamailio_icscf/kamailio_icscf.cfg, line 94, colu mn 12-19: failed to load module
scscf | 0(38) ERROR: <core> [core/sr_module.c:527]: ksr_locate_module(): could not find module <maxfwd> in </usr/lib64/kamailio/modules_k/:/usr/ lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux-gnu/kamailio/modules/:/usr/local/lib64/kamailio/
and there is docker file that Im using:
FROM ubuntu:focal
ENV DEBIAN_FRONTEND=noninteractive
# Install updates and dependencies
RUN apt-get update && \
apt-get -y install mysql-server tcpdump screen tmux ntp ntpdate git-core dkms \
gcc flex bison libmysqlclient-dev make libssl-dev libcurl4-openssl-dev \
libxml2-dev libpcre2-dev bash-completion g++ autoconf libmnl-dev \
libsctp-dev libradcli-dev libradcli4 libjson-c-dev pkg-config iproute2 net-tools \
iputils-ping
# Fetch Kamailio code (branch 5.3)
RUN mkdir -p /usr/local/src/ && cd /usr/local/src/ && \
git clone https://github.com/kamailio/kamailio && \
cd kamailio && git checkout 5.8
# Build and Install Kamailio
RUN cd /usr/local/src/kamailio && make cfg
COPY modules.lst /usr/local/src/kamailio/src
RUN cd /usr/local/src/kamailio && \
make -j`nproc` Q=0 all | tee make_all.txt && \
make install | tee make_install.txt && \
ldconfig
COPY kamailio_init.sh /
CMD /kamailio_init.sh
Hi all,
i have a problem with my Kamailio SIP Server.
I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN.
The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :(
I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it.
Can anyone help me to understand why? I can copy and paste here all the config files if you need it.
Thank you so much in advance
Christian
Hi Gang
Somehow I don't get my head around NAT Flags and the nathelper module
https://www.kamailio.org/docs/modules/5.7.x/modules/nathelper.html
In the examples I found, there is: FLT_NATS and FLB_NATB
If I got it right, FLB_NATB is a branch flag, which shall indicate that
the device is 'B'ehind NAT, right?
It is being set, when FLT_NATS is set:
if(isflagset(FLT_NATS)) {
setbflag(FLB_NATB);
}
But when should FLT_NATS be set and what is it's meaning? The examples I
found don't tell me this.
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G - Leiter Commerce Kunden
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
______________________________________________________
Hello,
We have a production Kamailio 5.2.7 server which has suddenly started
reporting spikes in the shared memory use. When looking into this, we
noticed that at the same time as the memory spikes Kamailio also logs a
huge amount of errors like the following.
Can anyone help with an explanation for what's going on? The xx.xx.52.202
address is the Kamailio server itself. We record historical numbers of
network connections for the server overall, and that doesn't show any
increase in the number of TCP connections, either at the time of the spikes
or as a general trend.
Thanks in advance!
Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: <core>
[core/tcp_main.c:2703]: tcpconn_1st_send(): connect xx.xx.52.202:42726
failed (RST) Connection refused
Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: <core>
[core/tcp_main.c:2711]: tcpconn_1st_send(): xx.xx.52.202:42726: connect &
send for 0x14bbf44c9f40 failed: Connection refused (111)
Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: tm
[../../core/forward.h:293]: msg_send_buffer(): tcp_send failed
Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: WARNING: <core>
[core/tcp_main.c:1149]: tcp_do_connect(): xx.xx.52.202:42726: could not
find corresponding listening socket for xx.xx.52.202, using default...
--
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
Hi everyone,
New user who is trying to make an outbound proxy to route a lot of freepbx
boxes for outbound calls.
I got this setup and working using dsiprouter. Calls come in from freepbx
trunk using outbound proxy, authenticated using IP auth, and then gets
signed with our stirshaken cert, then routes out to our carriers.
Interested in using plain kamailio. Got everything compiled and
libstirshaken compiled.
Just wanted to see if someone could point me in the right direction for the
cfg file.
I assume I would match how I have it in dsiprouter. That config has a lot
of things I'm not using and having a hard time deciphering it.
My thoughts on what I need:
MySQL database. Have a table for ip auth, a table for voice carriers, and
a table for 911 carriers.
Invite comes in, it checks ip auth table and responds with 403 if it's not
on the list. If it is, proceed to stir shaken signing, and then route the
call? I assume I will need to have some logic to tell voice calls and
emergency calls.
I got plain old routing/forwarding working and now want to focus on locking
it down.
Any thoughts, examples configs, anything would be awesome.
I am hoping to get this working and then having a primary/standby or even
load balancing them but im in the very early stages of trying to get my
head around Kamailio.
Thanks for any insights!
Hi List
I have stumbled over this challenge:
I have a kamailio server without dialog module. Only tmx module, acting
as registrar / rtpengine host.
If the call is routed over that server once, this works fine.
If the call is routed more than once (call from one location to another
on same registrar, call forwarding) then to correctly work, I would need
to engage rtpengine in a way to tell rtpengine to handle both call legs
separately.
By default, rtpengine identifies one call by Call-ID,FromTag,ToTag so
it considers all invocations to target one instance and on the 2nd
invocations it replaces the source rtp ip by it's own ip killing audio.
Using 'loop-protect' to prevent the 2nd invocation 'sort-of' works, but
not cleanly if I use different 'private' ip ranges on my voice core and
IC network to which CPE should not talk.
It is possible to pass a custom Call-ID to rtpengine. So this is how I
am considering finding a solution.
When passing the call-id to rtpengine I could append a leg identifier
to that call-id.
like $avp(rtp-callid) = $ci + $var(leg-id)
How could I reliably generate this leg-id / tell the legs apart?
Consider this situation:
leg 1 leg 2
CPE-A => Kam-Reg => Routing Core => Kam-Reg => CPE-B
Also I wonder if I can use an AVP at all? From Kamailio's Point of
view, the first invite being routed twice could be the same transaction
with same shared AVP, right?
Is there a way to pass some information via flags appended route-header?
Or maybe use the count of via or route header to tell the legs apart?
Any ideas very appreciated.
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G - Leiter Commerce Kunden
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
______________________________________________________
Hi everyone,
I am trying to assign environment variable as follows
listen=udp:0.0.0.0:5060 advertise $env(MY_IP):5060
Looks like using the environment variable as above is an invalid
configuration.
Is there a way to use IP from env var to advertise. Even better, is there a
way to use result in a stun query as an advertised address?
Thank you,
Pavan Kumar
Hi,
I see the documentation for the sqlite module includes a parameter to
set the journal_mode to wal, but I don't see any way to set
synchronous=normal. Is there a way to run arbitrary PRAGMA statements
upon opening the file or would I have to run it before each query?