I'm using opensips version 2.3 from rpm package on fedora. I've noticed
that if I use the wrong auth info for the mysql db and the connection fails
then opensips does a core dump in db_table_version in both the acc and
permissions modules. Probably more, I didn't bother going further down the
M4 will process the file before OpenSIPS runs and will not be changeable at
runtime. It sounds like that will not work for you if I am understanding
properly.
My first thought would be to use the shared variables from CFGUTILS [1]. They
can be accessed from inside the script, but can also be
I'm using a webserver response rather than cache_fetch but this may help.
#we got a valid response from the web server.
convert from string to json object
$json(resp-obj) := $avp(router-resp);
if( $avp(ok_log_this) == "true"
Daniel, you can find some v4-v6 examples here:
https://github.com/sippy/voiptests/blob/master/test_run.sh
${RTPPROXY} -p "${RTPP_PIDF}" -d dbug -f -s stdio: -s "${RTPP_SOCK_UDP}" \
-s "${RTPP_SOCK_CUNIX}" -s "${RTPP_SOCK_UNIX}" -s "${RTPP_SOCK_UDP6}" -s
"${RTPP_SOCK_TCP}" \
-s
Hi team,
I have another application storing the json encoded object in the redis
cache.
I read we can fetch using cache_fetch by passing the key. This is perfect
but I cant find a function to decode the json I have since my other
application is encoding the json object and putting into the
Hello Liviu,
Yes, that correct. Right now each node in cluster have own vip from
keepalived on LAN side.
The issue how to specify correct vip in case of failover, so if node 2
fail and node 2 vip was relocated to node 1. All sessions should be
process on node 1 and
append_hf("Path: \r\n");
Hi
I'm trying to configure RTPPROXY to bridge ipv4 and ipv6 networks, but
didn't find the proper way.
Supposing IPs "200.200.200.200" and "2607:3f00:2 " both on ETH0
interface.
Tried:
/bin/rtpproxy -F -l 200.200.200.200/2607:3f00:2
Got this error: Restarting rtpproxy: rtpproxy:
study, Fatma, study :)
btw, if you can't "find a softphone capable of being a presence user agent"
you may be in the wrong field of studies.
-giovanni
On 2 August 2018 at 15:52, Fatma Raissi wrote:
> Good morning,
>
>
> Thanks again for your answer.
> But I can't find a softphone capable of
The log suggests that the mediaproxy dispatcher is not replying in time, yet it
is not dead/frozen.
The timeout is 500ms and can be set through the mediaproxy_timeout module
parameter (see the mediaproxy module README for more info).
I suggest you increase the timeout until you don't see the
Hi all,
Just pushed a fix to include rtpengine table into the default create
tables, so `opensipsctldb create` will now create the rtpengine table too.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
Good morning,
Thanks again for your answer.
But I can't find a softphone capable of being a presence user agent.
Plus the presence information I need to publish is one variable which is
"workload" of the machine.
Here is the SIP message I am using and joined the configuration file. Maybe
you
Hi Volga,
If I understood your problem correctly, one idea would be to use m4 over
opensips.cfg and define a different MAIN_VIP variable for each of your
three servers:
append_hf("Path: \r\n");
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 09.07.2018 07:34,
Hey guys, sorry to keep insisting, im just very curious why its not finding
that TCP connection.
Nothing anywhere tells or yields an error.
What else can I look at?
On Mon, Jul 23, 2018 at 11:58 AM, Sebastian Sastre <
sastre.sebast...@gmail.com> wrote:
> I put a full debug on this paste
Hi all,
For the sake of completion, here is the commit fixing the issue:
https://github.com/OpenSIPS/opensips/commit/058cc22cb55dce9b890308b9f83a42a88691f2c8
Thank you Yuval for the report and for investigating this!
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
Be ause they have working presence client embedded, and you seems not be
able to model it in sipp.
Start with something known to work, softphones, trace the sip messages,
then (if needed) do the sipp xml modelization.
-giovanni
On Thu, Aug 2, 2018, 13:45 Fatma Raissi wrote:
> Good morning
Good morning Sir,
Thank you a lot for your answer.
But could you explain why would I use softphones while I have nothing to do
with voice or voice over IP.
Cordially,
Fatma RAISSI - ENIT Junior Entreprise
*Élève ingénieur en télécommunicationMembre d'honneurVice-Présidente du
mandat
Hi,
What you are trying to do it not logically correct ; even more, it is
not SIP wise correct (a proxy must not mess up with the final replies as
the end points will get de-syncronized .
This a well know race in SIP - caller canceling versus callee answering
the call. The RFC says that the
Use softphones instead of sipp
On Wed, Aug 1, 2018, 12:01 Fatma Raissi wrote:
> Good morning Everyone,
>
>
> I am using OpenSIPS as *presence server*. I need it just to accomplish
> very basic and simple presence server functions.
>
> Here is the purpose of my work:
>
> I have 3 machines P, A,
18 matches
Mail list logo