Hi all!
Is check config without restart possible?
--
WBR, Victor
JID: coyote(a)bks.tv
JID: coyote(a)bryansktel.ru
I use FREE operation system: 3.9.4-calculate GNU/Linux
Hi,
I want to do this:
2 hosts, with 2 running kamailios, every host has 1 IP address and hostA
has virtualIP assigned. kamailio should run on both hosts. I made a script
which can transfer virtual IP from hostA to hostB.
The problem is, I cannot tell kamailio to use virtualIP on hostB because,
when I set
listen=virtualIP:5060 - it cannot start because virtualIP is not present
listen=0.0.0.0:5060 and force_send_socket(virtualIP) - it does not work as
virtualIP is not in the listen list
Some hint? I could run kamailio automatically after chaning IP address, but
it requires time to load, so it will not be as quick as I change simply the
owner of virtual IP.
Thanks,
Mino
Hi! Has someone managed to get the Microsoft Lync client communicate with
the Lync Server through Kamailio? I tried but the Lync Server behaved
differently, refusing to send BENOTIFY requests. This caused presence not
to work properly, and IM not to work at all. I would be happy to hear if
someone had success with this.
Thanks.
Related to 'Doing automatic unregister when a WEBSOCKET connection is
closed' thread.
http://sip-router.1086192.n5.nabble.com/Doing-automatic-unregister-when-a-W…
Any hint to implement automatic call termination when websocket
connection is closed?
Some of my ideas:
1. send BYE message from UAC module. Looks like it is an unnecessary
complex & dirty solution.
2. Utilizing dialog module, but i'm not sure if dialog module can be
used to implement this.
Any better idea?
Thanks.
--
Iwan Budi Kusnanto
Hi,
I need to update some hash tables inside kamailio, but this update should
be triggered externally.
I know there is xmlrpc or xhttp, but I am just curious if there is some
easier way how to run this route. I have to run it instantly, so setting
some shv and run this route if new invite comes is not possible, nor timer.
Thanks,
Mino
Dear Daniel,
Thank you for your previous reply.
I am working on Kamailio (V 4.0) server to bring up VoIP features (Voice
call, SMS) on my Hand-held, Wi-Fi enabled devices (Which are not GSM
supportable).
1) I would like you to confirm that, is this Kamailio architecture supports
SMSing feature for end-to-end IP solutions? (without GSM module).
2) And i am also focusing on Kamailio integration with Asterisk server, so
is there any scope of get through my requirement (SMS on Kamailio server
through end-to-end IP) in this integration case?
I hope you could help me in getting clarity about this.
*Thank you and Best regards,*
*R Ravindra **Gowda*
*Engineer - Systems and Products*.
*Thrikasa Technologies*
*| Office: +91 40 23260434
| Mob: +918885284050
*
*| Fax:+91 40 23221045*
*| email: ravindra(a)thrikasa.in*
*| Web: www.thrikasa.in*
'If you don't give up,you cannot fail'.
The information contained in this communication is proprietary to Thrikasa
Technologies., and/or third parties, may contain classified or privileged
information, and is intended only for the use of the intended addressee
thereof. If you are not the intended addressee, please be aware that any
use, disclosure, distribution and/or copying of this communication is
strictly prohibited.If you receive this communication in error, please
notify the sender immediately and delete it from your computer.
On Sat, Jun 8, 2013 at 12:16 PM, Daniel-Constantin Mierla <miconda(a)gmail.com
> wrote:
Hello,
do not write to many mailing lists at once -- that should be done only for
projects announcements or asking opinions of both communities for changes
related to the project. Otherwise, send to sr-users if you have questions
about how to use kamailio and to sr-dev only for questions related to new
features development and unstable branch.
To help you, grab ngrep trace of registration and see who is sending 401.
If it is kamailio, then restart it with debug=3 in configuration file and
look at the logs to see why is not authenticated. If you fail to understand
the problem, send both ngrep output and syslog messages here.
Cheers,
Daniel
On 6/7/13 2:58 PM, Ravindra Gowda wrote:
Dear All,
I was successful in integrating kamailio 4.0.1 server with Asterisk 11.4.0
in Ubuntu 12.04 LTS version, by using the following link :
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb.
Both the servers are running parallely without any problems. And i choosen
siremis-4.0.0 to get the control panel of kamailio server and i got its GUI
(By following below link):
http://kb.asipto.com/siremis:install32x:main
But when i tried to register my VoIP phones with the Kamailio server, I am
getting the status code reply as *'401 Unauthorized* (0 bindings)'.
What could be the wrong? How can i solve it? And how can i retrieve my
Users (are which i had added to MySQL database) in SIP server (kamailio)
GUI?
Any help will greatly appreciate.
*Thank you and Best regards,*
*R Ravindra **Gowda*
*Engineer - Systems and Products*.
*Thrikasa Technologies*
*| Office: +91 40 23260434
| Mob: +918885284050
*
*| Fax:+91 40 23221045*
*| email: ravindra(a)thrikasa.in*
*| Web: www.thrikasa.in*
'If you don't give up,you cannot fail'.
The information contained in this communication is proprietary to Thrikasa
Technologies., and/or third parties, may contain classified or privileged
information, and is intended only for the use of the intended addressee
thereof. If you are not the intended addressee, please be aware that any
use, disclosure, distribution and/or copying of this communication is
strictly prohibited.If you receive this communication in error, please
notify the sender immediately and delete it from your computer.
_______________________________________________
sr-dev mailing listsr-dev@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-dev
--
Daniel-Constantin Mierla -
http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
Hi guys,
I need to implement a call forwarding (blind call forward) in a
kamailio.
Do you know if this is possible? There is a way that allow the
subscriber to configure it's own forwarding ?
Thanks in advance,
Lucas
Hello,
Kamailio SIP Server v4.0.2 stable release is out.
This is a maintenance release of the latest stable branch, 4.0, that
includes fixes since release of v4.0.0. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.0.0. Deployments running previous v4.x.x versions
are strongly recommended to be upgraded to v4.0.2.
For more details about version 4.0.2 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2013/06/kamailio-v4-0-2-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda