Hello,
I am having this problem on kamailio 1.5.2-tls compiled on Ubuntu 8.04,
1.5.1-tls ( compiled with no tls ) on Ubuntu 8.04 and with OpenSIPS
1.5.2-tls compilde on Ubuntu 8.04
I am trying to setup presence_dialoginfo with my Grandstreams, Snom and
Linksys. I have a 4 phones on the serve
2009/8/9 Sanyog Kale :
> sir
> i am working on project Build SIP-based VOIP Service With RADIUS AAA
> Using Kamailio (OpenSER) And FreeRadius as per tutorial given by you.
> but the problem is
>
> 1.i am not understanding how to intialize modules for freeradius
> 2.while testing radius server
I will have to generate an INVITE later on so i am using this topic to ask
you something. I'm working on having in one side a sequence (BYE +
(re-INVITE+ newSDP)). The new SDP will have some more media parameters than
the initial one.
I have been reading documentation, and it seems there is an
You can start with next document which is still good from architectural
point of view:
http://siprouter.teigre.com/doc/gettingstarted/
Main site for docs is here -- follow the links, the dokuwiki is one that
really worth:
http://www.kamailio.org/mos/view/Documentation-Repository/
To get more
You just have to read the documentation. No easy answer for any of your
questions.
Sanyog Kale wrote:
sir
i am working on project Build SIP-based VOIP Service With RADIUS
AAA Using Kamailio (OpenSER) And FreeRadius as per tutorial given by you.
but the problem is
1.i am not understan
sir
i am working on project Build SIP-based VOIP Service With RADIUS AAA
Using Kamailio (OpenSER) And FreeRadius as per tutorial given by you.
but the problem is
1.i am not understanding how to intialize modules for freeradius
2.while testing radius server where to make file named digest
3.t
