Hi,
I have the following /etc/apt/sources.list:
$ cat /etc/apt/sources.list
deb http://ftp.debian.org/debian squeeze main contrib non-free
deb-src http://ftp.debian.org/debian squeeze main contrib non-free
deb http://deb.kamailio.org/kamai
Hello,
I just pushed to remote GIT repository in master branch a bit of
refactoring about the states and ds_mark_dst().
Since with 3.2 seemed that it was lost capability to go inactive after a
certain number of failures (ds_probing_threshold), there is a new state
'trying' that can be used f
Hello,
I committed on master branch the code for support of dynamic codecs ids:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=511dc62e6a6ca74324f42b66a23bd9d80b377252
I tested and seemed ok, but could try many options, so if you can give
it a try as well would be good bef
Hi Daniel,
On 27/10/2011 15:57, Daniel-Constantin Mierla wrote:
> Hello,
>
> I just pushed to remote GIT repository in master branch a bit of
> refactoring about the states and ds_mark_dst().
Thanks, I will test the dev branch in a short while and get back to you.
>
> Since with 3.2 seemed that
Thank you for your quick reply. t_set_retr doesn't affects transaction
created by uac_req_send.
The second way proposed by you works great (tm settings for uac,
t_set_retr for t_relay).
Hello,
On 10/26/11 12:10 PM, Vitaliy wrote:
Hello,
How can i change T1 timeout for transaction created by
Hello,
On 10/27/11 5:30 PM, Asgaroth wrote:
Hi Daniel,
[...]
Since with 3.2 seemed that it was lost capability to go inactive after
a certain number of failures (ds_probing_threshold), there is a new
state 'trying' that can be used for it. Means that you can set a
destination in trying state c
On 10/27/11 6:15 PM, Vitaliy Aleksandrov wrote:
Thank you for your quick reply. t_set_retr doesn't affects transaction
created by uac_req_send.
The second way proposed by you works great (tm settings for uac,
t_set_retr for t_relay).
Just to be clear, you have tested t_set_retr only in rou
Hello,
you have to provide the sip trace taken on the sip server, in order to
see what is received and what is sent out by kamailio. Looks like the
one you pasted here is from client.
You can use ngrep on kamailio server:
ngrep -d any -qt -W byline port 5060
Also, the packets you pasted nex
Hello,
indeed, it looks like debs for 3.2 are not yet generated -- the apt repo
for it has not debs. Jon (cc-ed) was taking care of it, afaik, he is
traveling these days, so it may take a bit until he can loot at it.
Meanwhile, if you have a debian system where you can install the
dependenci
Hi
I am running kamailio proxy (1.5) as an intermediate proxy, where all my SIP
signalling packets are passing through.
I want to limit maximum number of calls kamailio proxy server can handle
at a time.
How can I do this, please give me some pointer in this regard.
Regards
Austin
One correction. I am using Kamailio 3.1.5.
On Fri, Oct 28, 2011 at 8:31 AM, Austin Einter wrote:
> Hi
> I am running kamailio proxy (1.5) as an intermediate proxy, where all my
> SIP signalling packets are passing through.
> I want to limit maximum number of calls kamailio proxy server can hand
Investigate the "dialog" module.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppiness.
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web
Thanks Alex
Do you mean that I need to modify kamailio.cfg to have max calls limitation?
Regards
Kamal
On Fri, Oct 28, 2011 at 8:47 AM, Alex Balashov wrote:
> Investigate the "dialog" module.
>
> --
> This message was painstakingly thumbed out on my mobile, so apologies for
> brevity, errors, an
On 10/27/2011 11:26 PM, Austin Einter wrote:
Do you mean that I need to modify kamailio.cfg to have max calls
limitation?
Quitely so.
--
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://ww
Hi
I did change kamailio.cfg as below and getting some error.
*//Loaded dialog module*
*loadmodule "dialog.so"*
My route block looks as below.
*# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
route {*
*# per request initial checks
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