Hi!
I had some issues with fail2ban running on OpenSuSE (different
versions) when monitoring more than 1 log files. While tracking down
the problem I found other reports on the internet about the similar
problems. Eventually I found OSSEC from TrendMicro
(http://www.ossec.net/main/downloads/)
Hi Sebastien,
Is the 'mhomed' option set in your configuration file?
regards.
On Sun, Nov 7, 2010 at 2:53 PM, Sebastien CRUAUX scru...@halys.fr wrote:
I'm getting the same problem with RTP packets... Maybe some issue with
rtpproxy bridging mode ?
Le 05/11/2010 11:21, Vallimamod ABDULLAH
Good day list, from day one i have one unresolved question about Nat
traversal, i know it works, but i dont know how.
Here is the classic situation:
10 phones Nat B2BUA
Let's say they are the Aastra phones, local sip is 5060, and local rtp is
3000 by default. Nat is intellegent one, it's a
Hi Bill,
as you have a multi interface system, have you tried to enable the
mhomed global parameter?
http://www.opensips.org/Resources/DocsCoreFcn16#toc60
Regards,
Bogdan
Bill W. wrote:
As an update, the load balancer probe appears to use the ip address
specified by the first listen=
Hi Gustavo,
Take a look to http://www.opensips.org/Resources/Install .
Also check the OpenSIPS Virtual Machine, to see how a running system
looks like:
http://www.voice-system.ro/shortcuts::opensips_livedvd
Regards,
Bogdan
Gustavo Gomes wrote:
Hello, my name is Gustavo, and I am studying
Hi,
Have you tried :
exten = _VMR_.,n,Voicemail(${EXTEN:4...@{sipdomain},u)
**
Regards,
Bogdan
osiris123d wrote:
I have set up Asterisk to work with OpenSIPS so that instead of the context
for all OpenSIPS Subscribers being default it is their actual domain.
Following the OpenSIPS Tutorial
Hi Jayesh,
yes, you cannot initialized different vals for an AVP by using
indexes...Why ? it is simple - the values are a continuous array, and by
allowing init via indexes, you cannot guarantee this continuity (like
setting val index 0 and setting val index 3that is bogus).
Regards,
Saúl Ibarra Corretgé wrote:
On 11/03/2010 04:00 PM, Hung Nguyen wrote:
Hi all, thanks for reply.
I have tested with pike module. It is very simple.
--
modparam(pike, sampling_time_unit, 3)
modparam(pike, reqs_density_per_unit, 20)
if (method = 'REGISTER | OPTION | BYE') {
Hi,
strange if you do not have any errors :(
I just made a fix on both trunk and 1.6 to extend some checks in
flatstore and prevent crashing (even if the DB op will not be executed).
Could you update from SVN and see if stops crashing ?
Regards,
Bogdan
thrillerbee wrote:
Bogdan,
I am
Hi Bill,
What is the probing interval you configured ?
http://www.opensips.org/html/docs/modules/1.6.x/load_balancer.html#id250040
Could you test the attached patch ? (same as revision 7356 on trunk).
Regards,
Bogdan
Bill W. wrote:
Hello Bogdan,
This is how I mark the destination
Hello,
There is a new release of MediaProxy available, it contains a major bug fix for
a crash that appears under heavy load.
To upgrade your Debian unstable installation do:
sudo apt-get update
sudo apt-get install mediaproxy-dispatcher mediaproxy-relay
mediaproxy-web-sessions
I can't use {SIPDOMAIN} because the {SIPDOMAIN} variable is actually the IP
address of callers phone as it appears in the location table.
On a side note I was able to not use P-Asserted-Identity. because of a
different issue I learned about the uac_replace_to() function. I was able
to place
Thanks Bogdan!
Now I have one more rr-question.
ACK request received by opensips listen on X.X.X.X at port 5060
ACK sip:74951000...@x.x.x.x:5060 SIP/2.0.
Max-Forwards: 10.
Record-Route: sip:Y.Y.Y.Y;lr=on;ftag=2204003977
Via: SIP/2.0/UDP Y.Y.Y.Y;branch=z9hG4bK9a79.79e07dd4.2
Via: SIP/2.0/UDP
Hey Bogdan,
Looks like your patch is working. My probe interval was set to 15
seconds, but I increased it to 60 for testing.
The disabling, probing, and re-enabling appear to be working correctly now.
Thanks so much!
Bill
On 11/8/2010 7:24 AM, Bogdan-Andrei Iancu wrote:
Hi Bill,
What
Hey Bogdan,
I enabled the mhomed=1 parameter, and now I'm getting a bunch of errors
in the logs.
ERROR:tm:t_uac: no socket found
ERROR:load_balancer:lb_do_probing: probing failed
ERROR:core:get_out_socket: no socket found
ERROR:tm:uri2sock: no corresponding socket for af 2
Thoughts?
Bill
On
On 9/11/10 3:55 AM, osiris123d wrote:
I can't use {SIPDOMAIN} because the {SIPDOMAIN} variable is actually the IP
address of callers phone as it appears in the location table.
On a side note I was able to not use P-Asserted-Identity. because of a
different issue I learned about the
Hi all
i installed opensips and mediaproxy in server 1, ip address : 192.168.1.39
and cdrtool and freeradius in server 2, ip address : 192.168.1.42
the opensips, mediaproxy, freeradius starting are fine
i use the link below as a guide
http://cdrtool.ag-projects.com/browser/doc/INSTALL.txt
the
Hi Jose,
This option is not set in my config but actually I managed to figure out
things using the ie and ei flags in the
rtpproxy_offer/rtpproxy_answer fonctions.
Thanks for your help !
Best Regards,
Sebastien
Le 08/11/2010 10:16, jose luis millan a écrit :
Hi Sebastien,
Is the
Hello everybody,
I am facing an issue in sip message sequence while I log the siptrace using
siptrace module.
The issue is random, means, generally it is fine, but sometimes the SIP
messages are out of sequence i.e. ACK get logged before 200OK, even
sometimes INVITE comes after 100 Trying or 180
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