I'm trying to configure HOMER sip capture server to do some accounting
of the cases when "503 Service Unavailable" message arrives from
capture agents.
I've implemented the following to record the source IPs from which 503
message originates.
onreply_route {
..
On 03.09.2014 03:09, Muhammad Shahzad wrote:
> Thank you so much for your informative response.
>
> Yes the "peer" may be correct term in this sense as i am trying to
> identify "devices" (SIP UAs or Proxy) that are directly connected to
> Kamailio via SIP signalling (i.e. there is no other inte
Hi,
I am new to kamailio. i configure kmailio server and make call as
local extensions like 1001 and etc.
Now i want to know how to change the header name appear on softphone using
script in kamailo as the extension remain same as 1001.
what are the files i used to alter and where are to loca
Hi Senthil,
You can use record_route function to add Record_route header
in INVITE message( use rr.so module). This record_route header will become
Route header while sending BYE message from client. Suppose if you want add
directly Route header, then you can use Insert_hf func
please guide me, how i learn kamalio as i am new to kamailio.
suggest me some book or resources.
waiting.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailm
Hello,
Look at uac_replace_from() from uac module. Then sqlops module can be used
for loadin values from database.
Cheers,
Daniel
On Wednesday, September 3, 2014, Talha Bukhari
wrote:
> Hi,
> I am new to kamailio. i configure kmailio server and make call as
> local extensions like 1001 a
Hello,
The $si is the source ip of the sip packet. Bu in a rtimer route there is
no packet received from the network.
So your config is not going to work for what you want to do.
You can try to use mqueue to push sql queries from sip worker to rtimer as
you need to write something to database.
Hi,
I have setup kamailio 4.1.0 on an EC2 xlarge instance. The voice and video
calls seem to work well when both the devices are connected to the same
network, however, when one device connects to a different network (the two
devices now are on different networks), they are able to register on SIP