Hello,
Kamailio SIP Server v4.2.7 stable release is out.
This is a maintenance release of the previous stable branch, 4.2, that
includes fixes since release of v4.2.6. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.2.x. Deployments running previous v4.x.x versions
are strongly recommended to be upgraded to v4.2.7 (or to 4.3.x series).
For more details about version 4.2.7 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2015/12/kamailio-v4-2-7-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Note: the latest stable branch is 4.3, at this moment with its latest
release v4.3.4. See more details about it at:
* http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/miconda - http://www.linkedin.com/in/miconda
Hi everyone,
I'm trying to make kamailio works as a SBC in front of Asterisk. I have read the tutorial http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and it is close to what I'm looking for but still a little different. I want don't want kamailio to do auth and I want it to be a media gateway. The media and SIP path have to look like that:
Phone 1 ----- kamailio -----Asterisk ---- Kamailio ---- Phone 2
First I have add an outboundproxy field in the Asterisk configuration to make all SIP messages from Asterisk passe through Kamailio. Then I started a minimal conf for Kamailio and it works at least for the SIP part. Call can be established perfectly.
However the RTP proxy part is a mess. Kamailio changes the IP address in the sdp part only for INVITE messages and not for the 200 OK responses, so there is only one side of the media passing through Kamailio, the other side is sent directly to asterisk. But even for the half part rtpproxy craches with a segfault and Kamailio doesn't transmit this media.
I have been searching some helps on google but nothing so far :/ I hope that someone here could help me.
-----------------------------------------------
#!KAMAILIO
####### Defined Values #########
#!define ASTERISK 192.168.51.43
####### Global Parameters #########
#!ifdef WITH_DEBUG
debug=4
log_stderror=no
#!else
debug=2
log_stderror=no
#!endif
memdbg=5
memlog=5
log_facility=LOG_LOCAL0
fork=yes
children=2
auto_aliases=no
# to avoid simple attacks, use unconventional port
port=5060
####### Modules Section ########
mpath="/usr/local/lib/kamailio/modules"
loadmodule "db_text.so"
loadmodule "mi_fifo.so"
loadmodule "kex.so"
loadmodule "sl.so"
loadmodule "rr.so"
loadmodule "pv.so"
loadmodule "maxfwd.so"
loadmodule "textops.so"
loadmodule "siputils.so"
loadmodule "xlog.so"
loadmodule "sanity.so"
loadmodule "ctl.so"
loadmodule "cfg_rpc.so"
loadmodule "mi_rpc.so"
loadmodule "path.so"
loadmodule "rtpproxy.so"
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
# ----------------- setting module-specific parameters ---------------
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)
modparam("path", "use_received", 1)
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
#!endif
####### Routing Logic ########
route {
xlog("L-ALERT", "new message !!!!!!!!!!!!!!!");
if (!sanity_check()) {
exit;
}
if ( !mf_process_maxfwd_header("10") )
{
sl_send_reply("483","To Many Hops");
xlog("L_NOTICE", "$si $rm $ru -- too many hops\n");
exit;
}
if (af==INET)
{
force_rport();
if (src_ip != ASTERISK)
{
$du = "sip:192.168.51.43:5060";
add_path_received();
route(RTPPROXY);
record_route();
}
else
{
loose_route();
route(RTPPROXY);
record_route();
}
}
forward();
}
route[RTPPROXY] {
rtpproxy_manage("cwei");
}
onreply_route {
if (af==INET)
{
rtpproxy_manage("cwei");
}
return(1);
}
---------------------------------------------
Guillaume
Hello,
later this week, on Thursday, Dec 17, I am planning to release a minor
version based on branch 4.2, respectively 4.2.7 -- it's been a while
since v4.2.6.
As usual, if there are issues not yet reported or patches to be
backported, write to sr-dev mailing list.
If time allows, in the same day or next days, I plan to do the last
release of branch 4.1 to mark the end of official packaging in 4.1
series. At this moment, the last two stable branches are 4.3 and 4.2.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
Hello,
I use kamailio 4.3.2 with the presence module.
When the contact is able to manage its presence, No problem for me.
But when a contact doesn't manage it presence, i would like to know if it
is possible to generate a sip message (like as a simple presence message)
with kamailio?
The goal of that is to prevent my watcher when a "simple" contact (without
presence option) is registered and unregistered on my kamailio server.
Thank you for your help.
Best regards.
Loïc BOISSY
Hello,
based on the feedback after the proposal, there were no voices against
the new logo. Therefore, as of today, Kamailio project is officially
using the new logo -- see it at:
- http://www.kamailio.org/w/2015/12/new-logo-for-kamailio-project/
If you display one of the old Kamailio logos, we would encourage to use
the new one at your earliest convenience. That will help to propagate it
faster. Posting about the change on blogs, forums or social media
channels would be appreciated as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
Hello,
I am under the impression that the name dialog_ng creates confusion out
there and some people are using it instead of the classic dialog module.
Although it was started with goals of reworking dialog module with a
different concept (which was discussed mainly by some guys that
afterwards changed their job to non-voip area), dialog_ng ended up to be
tailored for IMS needs.
Probably we should do that refactoring of the dialog module, but
meanwhile dialog_ng doesn't refect that and some people are confused by
the current naming of the two modules.
Practically is more about convenience at this moment and if IMS
developers and users think it is not going to be a big overhead for
their deployments to be upgraded, I can take care to rename it. So,
while general opinion matters, I think we should see first what IMS devs
prefer.
I am personally not affected that much, so I am fine to keep it like it
is now -- in that case, proper notes should be added to documentation,
stating that dialog_ng must be used only for IMS (or when the config
writer knows very well what she/he is doing).
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu
Hello again
First of all I wanted to ask if someone ever implemented that
http://kamailio.org/docs/modules/4.3.x/modules/imc.html with WebRTC
Second question is - I don't understand the logic. In description is said:
Handles Message method.It detects if the body of the message is a
conference command.If so it executes it, otherwise it sends the message to
all the members in the room.
But why in example (well however it has broken syntax) to IMC manager are
sent only messages from chat-rooms? How message from client can possibly
reach imc-manager then?
Also - when I send message with body "1111111" from user 1001 and
imc_manager catches it - I receive 500 command error. Why? :/
All that is working on top of sipjs-demo.
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
Hi,
It is not very clear to me how to handle serial forking scenarios with
rtpengine where there are multiple RTP listeners attached to multiple
network interfaces.
That is to say, given:
rtpengine ... -i net1/xxx.xxx.xxx.xxx -i net2/yyy.yyy.yyy.yyy
1. I attempt to reach SIP gateway 1 via net1, calling rtpengine_manage()
in the process:
rtpengine_manage("replace-origin replace-session-connection
ICE=remove direction=$avp(rtp_src) direction=$avp(rtp_dest)");
2. The call to gateway 1 fails, and I roll over to gateway 2, reached
via net2, with dispatcher.
I have some logic that checks how $rd has changed and sets
$avp(rtp_dest) accordingly in the dispatcher failure_route[].
3. In that failure_route[], I call:
rtpengine_manage("replace-origin replace-session-connection
ICE=remove direction=$avp(rtp_src) direction=$avp(rtp_dest)");
4. When looking at the SDP for the INVITE on branch #2 going out to
gateway 2, it does not appear to utilise media endpoint yyy.yyy.yyy.yyy,
but rather continues to represent itself outwardly as xxx.xxx.xxx.xxx,
even to gateway #2.
I can't figure out what I'm doing wrong here, and the 'rtpengine'
documentation only has one mention of serial forking, in the context of
the discussion of the 'via-branch' parameter. As far as I can tell, this
is not relevant to my use-case; I am not concerned about deleting the
whole call from rtpengine, but rather about updating the SDP offer
consonantly with serial forking to gateway #2.
Any help would be deeply appreciated!
Thanks,
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Hello everyone!
I need to relay 486/408/... other failure codes back to initial INVITE
server. Here
http://lists.sip-router.org/pipermail/sr-users/2010-November/066382.html is
recommended just to exit failure_route, but that didn't work for me. I need
that to let Freeswitch know which cause has ended the call. Now Kamailio
just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me
how to achieve that?
Thanks in advance
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
