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 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
______________________________________________________
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,
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?
Hi everyone,
Here is my setup
SIP Client (MacBook) <=> Kamailio (AWS) <=> Freeswitch (AWS)
When I initiate a call from Client, audio won't be sent out, but the client
can receive audio. (When the client receives the call everything works
fine). I started to trace the SIP message using tdpdump. And here is an
interesting thing.
Kamailio sends the following SDP message while sending 200 OK
v=0
o=FreeSWITCH 1714278184 1714278185 IN IP4 *<freeswitch-ip-address>*
s=FreeSWITCH
c=IN IP4 *<freeswitch-ip-address>*
t=0 0
m=audio 27644 RTP/AVP 120 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
But on my computer where SIP client is running I receive following message
v=0
o=FreeSWITCH 1714278184 1714278185 IN IP4 *<kamailio-ip-address>*
s=FreeSWITCH
c=IN IP4 *<kamailio-ip-address>*
t=0 0
m=audio 27644 RTP/AVP 120 101
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
Somehow, somewhere, somebody is modifying the IP address in the SDP
message, hence the SIP client is sending audio to "Kamailio" instead of
"Freeswitch" server.
My initial guess was my router might be modifying it. After checking the
router settings I found SIP ALG is enabled. I disabled it, yet no use.
I changed my internet connection to a different provider and things started
working. SDP is not getting modified.
Have you ever faced such an issue where someone/something is modifying the
SDP messages. How did you debug to figure out which component was modifying
it? And common fixes for such weird situations?
Thanks and Regards,
Pavan Kumar
Hi All,
If run Kamailio in Active/Passive mode lets say there are 100 transactions going on Active node and then Active went down, now i want passive to be aware of all those ongoing transactions so that it can handle ongoing calls transactions smoothly so how to store them in db? Is there any way to do so? i know there is module dialog for tracking calls etc but i just want to handle transactions.
Thanks in advance.
Ram
Hello everyone,
While going through the WebRTC example configuration (cfg) to better
understand and implement it in my setup, I came across a potential issue in
the code block. The code in question can be found at this link:
https://github.com/kamailio/kamailio/blob/465994de2859c7863b4cef8457be0a207…
In the code, there is an "if else" block, "if(loose_route())" that seems to
be incorrect. It appears that the "if" block will always evaluate to true,
given that "loose_route" never returns 0. According to the documentation,
the return values of "loose_route" are specified as 1, 2, -1, -2, -3.
Therefore, it seems the "if" block is unnecessary, and the "else" block
will never be executed. Could someone please confirm this observation and,
if accurate, suggest a correction for the code?
Thank you for your attention to this matter.
Best regards,
Pavan Kumar
Hello list,
I'm using Kamailio to balance SIP request between two Asterisk. I have a
problem with Parking calls using Asterisk Application. When a user
parking a call, some times can't pick up parked call because the call to
pick up the Call go to the other Asterisk.
I'm using the 1 algorithm "hash over from URI" without success.
Any hint?
--
---
I'm SoCIaL, MayBe