HI Team,
I use tls and xhttp module to enable websocket interface for webrtc client
to connect.
I wanted to add "Strict-Transport-Security" header to xhttp response to add
security. But currently I don't see a way to add it anywhere.
I can see only server_header as an option, But this will get added to SIP
messages as well.
It 'd be great if anybody has a suggestion around this.
Thanks
Varunan
El Tue, 18 Jun 2024 13:54:41 -0500
Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió:
> Just want to add that exec is heavy and slow. I would not recommend it.
>
> Is there a reason you want to do this over http-ifying your script and
> using async?
>
>
Not really. The scripts are provided. Maybe I'm against "http all the
things" which seems to be what everyone does nowadays.
Why would http client be lighter than executing a script? That would be a
design problem, right?
--
PekePBX, the multitenant PBX solution
https://pekepbx.com
Hi all
I've been working with async http client and async db queries in the past but
now I have to execute scripts and store the return values in avps.
Since I have no experience with that, I wonder how to achieve concurrency with
that scenario. Is there a way to exec async or suspend until the exec has
finished and resume later?
cheers,
Jon
--
PekePBX, the multitenant PBX solution
https://pekepbx.com
Hello,
I am doing BLF but I want different users to be able to setup different
dialog info initial states. My question is if there's an already
implemented way for doing that?
I've looked over docs and C code so far and this is what I've found:
1. modparam("presence_dialoginfo", "force_dummy_dialog", 1) => will force sending initial NOTIFY with dialog state "terminated"
2. Function calls: dlginfo_agg_nbody()->dlginfo_agg_nbody_empty() which
will build the pre-defined xml body with that "terminated" state. If not
implemented so far, maybe this can be used to set a different initial
state for a user (e.g. export a cfg function or take that info from DB?)
Any opinions on this?
Thank you,
Stefan
When I start Kamailio master version that has been build on Debian 12
with default-libmysqlclient-dev depending on libmariadb-dev-compat, I
get several warnings like this:
2024-06-17T14:48:27.702827+03:00 siika /usr/bin/sip-proxy[297246]: WARNING: db_mysql [km_my_con.c:187]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring
How can I prevent them?
-- Juha
Hello kamailio lovers,
I am currently working on a Kamailio configuration where I need to check the source port of incoming SIP messages. Initially, I used the $sp pseudo-variable to get the source port, which worked perfectly while testing with SIPp.
Here is the snippets code that I use
"
if ($sp != 5080 && $sp != 8080 && $sp != 5060){
sl_send_reply(403, "forbidden");
xlog("this $rm received, is forbidden\n");
exit;
}
"
However, I encountered an issue when I switched to using sipsak for testing.
It seems that sipsak uses a randomly assigned port for the actual network connection, even though it sets the source port in the SIP headers (e.g., Via, To and Contact headers) to a specified port using the -l option.
Here is the sipsak command that I use "sipsak -vv -s sip:jack@192.168.1.2:5060-l 5080"
My goal is to have Kamailio check the source port as specified in the application layer, rather than the transport layer port (i.e., the ephemeral port assigned by the OS randomly like 54691, 33017...).
Could anyone guide me on how to properly configure Kamailio to extract and check the port from the application layer or SIP headers, or am I doing this in the wrong way? Any advice or examples would be greatly appreciated.
"I am not a native english speak, so I use chatGPT to make my questions clear"
Thank you!
Hello,
I am using topos module to hide Record-Route and Route header as per the
requirement from carrier. But the topos module hides, contact, via and
routing headers also. Due to this reason when I am getting bye from the
carrier I am not able to route the bye back to the agent.
Bye from Agent to carrier is working fine.
Bye from carrier to my internal network interface is not routing to client
through public interface. I assume this may be because Via / Contact header
has been hidden and changed by the topos module with the invite
please help me for this
Thanks
Arun
Hello,
Kamailio SIP Server v5.8.2 stable release is out.
This is a maintenance release of the latest stable branch, 5.8, that
includes fixes since the release of v5.8.1. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.8.x. Deployments running previous v5.8.x
versions are strongly recommended to be upgraded to v5.8.2.
For more details about version 5.8.2 (including links and guidelines to
download the tarball or from GIT repository), visit:
 * https://www.kamailio.org/w/2024/06/kamailio-v5-8-2-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Hello,
I am considering to release Kamailio v5.8.2 (out of branch 5.8) on
Wednesday,
June 12, 2024. If anyone is aware of issues not yet on the bug tracker,
report them there asap in order to have a better chance to be fixed.
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Hi,
In the "Kamailio v5.8.0 Release Notes" I see: "SIP overload control support
for server (RFC7339)". Where can I find some more details about the feature?
Also, the similar question about the new "influxdbc" module. What are the
possible use cases for this module?
Thank you,
Leonid Fainshtein
Xorcom Ltd