Hi, I'm setting up shared user location for 2 instances of kamailio via
dmq_usrloc.
Everything seems fine except my setup is multi-domains.
The issue arises when the producer node produces the Contact of domain X,
Contact of domain Y separately, but the consumer node stores both as the
Contact of domain Y (seen in kamcmd ul.dump), in which Y is set in
usrloc_domain.
modparam("dmq_usrloc", "usrloc_domain", "{{domain_Y}}")
modparam("usrloc", "use_domain", 1)
====
I had a quick ngrep and see the domain was put in the "aor" key of the body
in KMQ message for usrloc
{"action":1,"aor":"test_username_0dppxcl@$domain_X"}
But I'm not sure how to make my consumer node to store the contact in
appropriate domain.
Any help will be appreciated.
Thanks and regards,
Loi Dang Thanh
Phone : +84. 774.735.448
Email : loi.dangthanh(a)gmail.com
Issue : Not getting relay of ACK and BYE to the next hop after the call is
answered
my Scenario : Asterisk ------->kamailio sip proxy------------------->
carrier (outgoing call)
My carrier is not allowed to send the SIP packet with Record-Route header.
So that I have removed record_route(). After that the call is getting
connected.
I am getting 200 OK (SDP) from carrier side and forwarded that to the
Asterisk on the other side. As a response I am getting ACK from asterisk.
But the kamailio is not forwarding the ACK to the carrier side. I
understood this is because the record-route is not there. The same thing is
happening for BYE also. The Bye is not forwarding to carrier side.
Kindly suggest me a solution for this for relaying ACK and bye without
Record-Route in kamailio
Bellow is the 200 OK SDP I am sending back to asterisk
2024/06/02 10:27:04.756610 103.155.114.101:5060 -> 103.182.153.113:5060
SIP/2.0 200 OK
Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2
Via: SIP/2.0/UDP 103.182.153.113:5060
;received=103.182.153.113;rport=5060;branch=z9hG4bKPj463fedcf-6258-4655-a53d-58ea57f144af
To: <sip:09496381412@103.155.114.101
>;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
From: <sip:917946357720@gaesip.teleforce.in
>;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
CSeq: 22823 INVITE
Allow:
INVITE,BYE,REGISTER,ACK,OPTIONS,CANCEL,SUBSCRIBE,NOTIFY,INFO,REFER,UPDATE
Contact: <sip:lucentNGFS-115804@10.5.110.117:5060
;alias=10.5.110.117~5060~1;x-afi=11>
Content-Type: application/sdp
Session-Expires: 7200;refresher=uas
Supported: timer
Content-Length: 248
v=0
o=LucentPCSF 130227946 130227946 IN IP4 103.155.114.101
s=-
c=IN IP4 103.155.114.101
t=0 0
m=audio 12806 RTP/AVP 8 101
-------------------------------------------------------------------------------------------------------
The ACK I am getting back from asterisk is
2024/06/02 10:27:04.760392 103.182.153.113:5060 -> 103.155.114.101:5060
ACK sip:lucentNGFS-115804@103.155.114.101:5060;alias=10.5.110.117~5060~1;x-afi=11
SIP/2.0
Via: SIP/2.0/UDP 103.182.153.113:5060
;rport;branch=z9hG4bKPjb7299c25-13d3-484e-8e78-e7c83620edce
From: <sip:917946357720@gaesip.teleforce.in
>;tag=8dba28f5-d250-48c6-99fc-35d84590cfc1
To: <sip:09496381412@103.155.114.101
>;tag=6541fd97-665bfb9a2aea9a78-gm-po-lucentPCSF-109109
Call-ID: 1d2897fc-8ae2-4835-b7ac-a96cc28adcc2
CSeq: 22823 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.13.0
Content-Length: 0
Thanks
Arun
Hello all!
I am struggling on compiling StriShaken module on RHEL 9.2.
So far, the module was compiled as well as libstirshaken.
Below are the steps used to compile (as sudo) both library and module (I
hope this may help someone) and, of course, correct me if I am wrong in any
step or if there is a better way.
Also, note that the RHEL 9,2 is in a VM environment without access to the
WWW, instead it uses Red Hat Satellite to download packages (via Yum or
DNF), if available.
Lib LibKS
download libks from https://github.com/signalwire/libks
unzip
move to dir created by Unzip
run
yum groupinstall "Development Tools"
dnf install libuuid-devel libatomic openssl-devel
cmake .
make
make install
cp /usr/lib/pkgconfig/libks2.pc /usr/lib64/pkgconfig/.
For module LibStirShaken:
download https://github.com/signalwire/libstirshaken
unzip
move to dir
if using OpenSSL3.0 or + edit file configure.ac and add after line 28:
if test x$HAVE_OPENSSL = x1; then
openssl_CFLAGS="$openssl_CFLAGS -DHAVE_OPENSSL";
/* PATCH FOR OPENSSL3 */
AC_MSG_CHECKING([for OpenSSL >= 3.0.0])
AC_COMPILE_IFELSE([AC_LANG_PROGRAM([[
#include <openssl/opensslv.h>
#if OPENSSL_VERSION_PREREQ(3,0)
#error "you_have_version_3"
#endif
]], [[]])],
[ AC_MSG_RESULT([no]) ],
[ AC_MSG_RESULT([yes]);
AC_DEFINE(OPENSSL_SUPPRESS_DEPRECATED, 1, [disable openssl
deprecated-function warnings]) ])
/* END OF PATCH */
else
AC_MSG_ERROR([OpenSSL >= 1.0.1e and associated developement headers
required])
fi
run
configure
make
sudo make install
next, move to dir /root/kamailio/kamailio-5.7.4/src/modules/stirshaken
cd /root/kamailio/kamailio-5.7.4/src/modules/stirshaken
ln -s /root/stirshakenlib/libks-master/src/include/libks libks
make
make install
All seem to compile OK , *but* when starting Kamailio I get the following
errors:
May 29 17:42:02 kamailio1 kamailio[248991]: ERROR: <core>
[core/sr_module.c:599]: ksr_load_module(): could not open module
</usr/local/lib64/kamailio/modules/stirshaken.so>: libstirshaken.so.1:
cannot open shared object file: No such file or directory
May 29 17:42:02 kamailio1 kamailio[248991]: CRITICAL: <core>
[core/cfg.y:3915]: yyerror_at(): parse error in config file
/usr/local/etc/kamailio/kamailio_stir_shaken_mod.cfg, line 106, column
12-26: failed to load module
I haven't found a solution for this.
Can anyone help?
Thanks in advance!
*Sérgio Charrua*
