Hello,
I am proposing a new IRC meeting to discuss the current major issues and
the plans for next Kamailio release, on Wednesday, April 20, 2016, with
alternative for next day, April 21. If many developers are not
available, we can postpone it to another date in the near future (make
proposals if that is the case for you).
I created a wiki page for it:
* https://www.kamailio.org/wiki/devel/irc-meetings/2016a
Add there the topics that you want to be discussed.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
Hi There,
I'm using dmq and dmq_usrloc between 2 kamailio(version 4.4.0) registrar
servers.
I'm registering 15000 subscribers with sipp and everything is ok, the
registration address of record are shared between both registrar servers.
The problem here is when I restart one of the registrar server and then
the other registrar server starts synchronization to restarted server.
The error that i see on this process is the following shown bellow:
ERROR: dmq_usrloc [usrloc_sync.c:75]: add_contact(): Invalid cseq
But all the contacts are synchronized, what can cause this error? and what
is the impact on normal behavior of kamailio?
Best regards
--
José Seabra
Hello,
Using kazoo only to connect to amqp gives an error if the db_url is not
specified. If db_url is specified it tries to do presence related stuff
even if not intended. The solution was to use:
modparam("kazoo", "pua_mode", 0) which I found here:
http://lists.sip-router.org/pipermail/sr-users/2015-September/090092.html
This is not documented in the kazoo module. Would be great if this is added
in the module documentation to simplify the use of module.
Thanks,
- Jayesh
A KA B
INVITE ->
200 OK(INVITE) <-
ACK is lost
BYE <-
200 OK(BYE) ->
200 OK(INVITE) <-
200 OK(INVITE) <-
200 OK(INVITE) <-
ACK is lost AND A sent BYE cmd,B do not recv ACK and is also Retransmission
200 OK, How can I distinguish this 200 OK and drop it?
thaks
Time is passing and Kamailio World Conference 2016 is approaching at
fast pace – only four weeks left till the start of the event!
The schedule [1] is pretty much nailed down, with some adjustments still
expected to happen. The event starts like the past edition with a half a
day of technical workshops, followed by two full conference days.
A larger group of speakers [2] is participating to this edition. There
was a big number of speaking proposals and we wanted to highlight more
of the people that had a relevant contribution to the evolution of the
project. To accommodate properly, two more discussion panels were added,
keeping also the classic VUC panel.
The topics cover many of the interesting aspects of real time
communications, from security and scalability to WebRTC and VoLTE,
touching Kamailio and other open source projects like Asterisk or
FreeSwitch.
More details can be found on the website of the event:
- https://www.kamailioworld.com
Don’t forget that this year Kamailio celebrates 15 years of development,
the party is at Kamailio World!
We expect to fill the capacity of the conference room, if you haven’t
registered yet and plan to attend, do it as soon as possible to secure
your seat! [3]
Many thanks to our sponsors [4] that made possible this event: FhG
Fokus, Asipto, Sipwise, Matrix.org, Sipgate, Simwood, NG Voice, Digium,
VoiceTel, Evariste Systems, Core Network Dynamics, Pascom, Didx.net.
Thank you for flying Kamailio and looking forward to meeting many of you
at Kamailio World 2016!
Daniel
[1] https://www.kamailioworld.com/k04/schedule/
[2] https://www.kamailioworld.com/k04/speakers/
[3] https://www.kamailioworld.com/k04/registration/
[4] https://www.kamailioworld.com/k04/sponsors/
/
/
--
Daniel-Constantin Mierla
http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
Hello:
I have kamailio with drouting and I want to pass only the RTP trafic to a
FREESWITCH o what ever you think is better to manage de RTP traffic. What
opcions do I have? I am a little bit confused using WITH_FREESWITCH.
I will appreciate your help,
Thanks
I may have stumbled across another bug with this module. I believe the htable.dump method isn't correctly recognizing that a table name is being provided:
[root@server ~]# curl -s --header 'Content-Type: application/json' --data-binary '{"jsonrpc": "2.0", "method": "htable.dump", "name": "callstats"}' http://1.2.3.4:5060/jsonrpc
{"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}}
Debug log:
2016-04-19T17:42:30.046892+00:00 server /usr/local/sbin/kamailio[21634]: DEBUG: <core> [tcp_main.c:2465]: tcpconn_do_send(): buf=#012HTTP/1.1 500 No htable name given#015#012Sia: SIP/2.0/TCP 1.2.3.4:54912#015#012Content-Type: application/json#015#012Server: kamailio (4.4.0-rc1 (x86_64/linux))#015#012Content-Length: 69#015#012#015#012{"jsonrpc":"2.0","error":{"code":-32000,"message":"Execution Error"}}
Brooks Bridges | Sr. Voice Services Engineer
O1 Communications
5190 Golden Foothill Pkwy
El Dorado Hills, CA 95762
office: 916.235.2097 | main: 888.444.1111, Option 2
email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>
What is this? It's referenced here: http://kamailio.org/docs/modules/stable/modules/htable.html#idp1912904 but I cannot find it anywhere else.
The only 2 places I find any reference to it at all in the entire codebase is here:
modules/htable/doc/htable_admin.xml: <emphasis>$shtval(htable=>key)</emphasis>
modules/htable/README: * $shtval(htable=>key)
Brooks Bridges | Sr. Voice Services Engineer
O1 Communications
5190 Golden Foothill Pkwy
El Dorado Hills, CA 95762
office: 916.235.2097 | main: 888.444.1111, Option 2
email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>
Hi all,
I'm trying to make a blind call forward to "1002" if a UA calls "1" into Kamailio. At the moment I have the following config:
if ( $rU=~"^[1]$" && src_ip==$sel(cfg_get.pstn.gw_ip) ) {
sl_send_reply("181", "Redirecting");
$ru = "sip:1002@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port) + ";user=phone";
force_send_socket(udp:MY_EXTERN_IP:5060);
route(NORMAL_RELAY); #Just t_on_branch/reply/failure
if (!t_relay()) {
sl_reply_error();
}
exit;
}
The calling UA is from the PSTN as well as the called UA. Problem right now is that the PSTN's anti-Loop protection jumps in and rejects the INVITE to 1002.
Question: Is it possible to tell the calling UA "Hey wait, 181..." with Kamailio, then build seperately an independent Call Leg (with new CallID?) from Kamailio to UA "1002" and finally to put them together?
Any help is very much appreciated even if they are just guides/examples. I'm using Kamailio 4.3.5 on CentOS 7.2.
Very Respectfully
Dimitry Nagorny
Trainee