Hi all
I’m trying to create a relatively simple setup with Kamailio dual homed on public/private ip and asterisk on private ip only. The idea is load balance / fail over asterisk boxes.
Following the real-time tutorial I have clients registering with Kamailio, Kamailio registering on clients behalf with asterisk as well as invites going through.
However what I’m seeing is that when an invite occurs asterisk offers media on its private ip, as it would. However this is making its way through Kamailio all the way to the client.
After a bit of searching all I can find is people trying to get it working and failing, or putting asterisk on public IP.
So questions - am I doing this completely the wrong way? Should Kamailio alter the media ip of asterisk on the way through or do I need to do that by hand? Surely someone somewhere has a write up on this already :-)
Thanks
Mark
Hello,
I have a problem today, It's strange for me.
Suppose we have this senario:
uac1------->SEMS(mo profile)------->Kamailio-------->SEMS(mt
profile)---------->uac2
In above topology, we have two interfaces(intern,extern) for SEMS, and
just used as SBC (sbc application).
if i used port=5060 as external port, every things is right and log
file is like this:
[#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd
msg via UDP from 89.165.117.125:42411 vv
--++--
REGISTER sip:kava.shatel.ir;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
89.165.117.125:42411;branch=z9hG4bK-d8754z-021bd8b61efc7ac0-1---d8754z-
Max-Forwards: 70
Contact: <sip:4000@
89.165.117.125:42411;rinstance=79011092e56e1a09;transport=UDP>
To: <sip:4000@kava.shatel.ir;transport=UDP>
From: <sip:4000@kava.shatel.ir;transport=UDP>;tag=82820e1f
Call-ID: Y2U4YThiYjEwNTUzMzliZTIwNWZkMDI3MTM4OTZlNWU.
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 0
but when i changed port=4080 for external port, The Via header and
contact header are changed to my public ip, like this:
[#7fed3f9f9700/32820] [run, udp_trsp.cpp:352] DEBUG: vv M [|] u recvd
msg via UDP from 89.165.117.125:42411 vv
--++--
REGISTER sip:kava.shatel.ir;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
172.1.1.125:42411;branch=z9hG4bK-d8754z-021bd8b61efc7ac0-1---d8754z-
Max-Forwards: 70
Contact: <sip:4000@172.1.1.125:42411;rinstance=79011092e56e1a09;transport=UDP>
To: <sip:4000@kava.shatel.ir;transport=UDP>
From: <sip:4000@kava.shatel.ir;transport=UDP>;tag=82820e1f
Call-ID: Y2U4YThiYjEwNTUzMzliZTIwNWZkMDI3MTM4OTZlNWU.
CSeq: 2 REGISTER
Expires: 3600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS,
INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.3.25608 r25552
Allow-Events: presence, kpml
Content-Length: 0
Why my public ip address is there here? I just changed external
port=4080 in sems server. In SDP protocol i have the same problem when
i changed external ip port in sems server.
Thanks.
--Mojtaba Esfandiari.S
Hello All,
I am currently using Kamailio and Asterisk on Centos 7 servers and trying to enable WebRTC jsSIP clients to be able to do Audio/Video calls with Provider Phones (Purple, Z, Sorenson, etc....), however, the providers do not have vp8 codecs (which is what the WebRTC clients use for Audio) so I believe I will need a media proxy server to resolve the video issues. My question is, can rtpproxy or rtpengine perform this transcoding? If so, and if rtpengine is the way to go, should I use Ubuntu for the rtpengine since it is the only one that seems to have a working installation?
Thank you,
-Steve
Hello,
I tried to install Kameilio on Ubuntu 16.04 from what explained in the wiki.
But when I execute the command to add your GPG key, I just get replied that the timeout occured.
What to do ?
Alain
Provenance : Courrier pour Windows 10
Hi,
My kamailio.log file is empty, following is my configuration:
In Kamailio.cfg-
log_facility=LOG_LOCAL0
#!ifdef WITH_DEBUG
debug=4
log_stderror=yes
#!else
debug=3
log_stderror=no
#!endif
Inside rsyslog.conf
# don't log messages with LOG_LOCAL0 in /var/log/syslog anymore
*.*;auth,authpriv.none,local0.none -/var/log/syslog
#
# log messages with LOG_LOCAL0 in /var/log/kamailio.log
local0.* -/var/log/kamailio.log
User has all the privileges to write in log file.
Regards,
Ashutosh Chaubey
Maxmind just announced EOL of legacy geolite databases used by geoip module,
no updates starting this April. See more at https://blog.maxmind.com/
2018/01/02/discontinuation-of-the-geolite-legacy-databases/
If you use kamailio's geoip module, I suggest to switch to geoip2 module which
uses Geolite2 databases and is a plug in replacement for geoip module.
Hi all,
When switching from kamailio 4.4.5 to kamailio 5.1.0 with the IMS installation in our lab, we detected three bugs, that lead to core dump.
We will issue three pull requests in the course of this afternoon.
KR
____
Christoph VALENTIN
Software Developer | R&D
Core / MGW + HLR + MSC
P +43 50 811 3785 | M +43 664 628 3785
christoph.valentin(a)kapsch.net<mailto:christoph.valentin@kapsch.net>
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Company Register at: Commercial Court Vienna FN 223804 z | Registered Office Vienna | www.kapsch.net<http://www.kapsch.net/>
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Hey everyone.
Right now i'm doing a redirect server and i'm trying to sinalize the
redirection using a 3xx reply. In order to do that, according the RFC I
need to add a Contact and Diversion header. How do i do that ? i'm creating
the reply with send_reply() but i can't seem to find a way to add these
parameters to the reply.
Thanks for the help.
Cheers