Happy to to report the max_load worked in our testing. Thanks,
On Tue, Sep 15, 2015 at 3:04 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> good to know that you found the issue and it was in the config file. Would
> be appreciated if you can report the results on testing
Hello all, we are working on a SIP solution using Kamailio.
We want to secure our base of user credentials even in case of attack on
the SIP server, and for that reason we plan to use diameter authentication
as described in RFC http://www.rfc-base.org/txt/rfc-4740.txt
Paragraph 6.2 describes a
Hi,
I have installed Siremis and modified it for public user
registration. I have added a new user. When I try to log in it opens
up a new (debug) window and prints "1". It doesn't allow me to log in.
But I can log in as admin;admin.
And siremis/user/register doesn't load properly. It redirects
Hi,
The way youve described it seems like you are not routing anything at all
to the asterisk. It all depends on your comfiguration on how you handled
the call. Somehow Ive a feeling that asterisk is used only for voicemail
and is called only once the B party is not found in lookup(location)
The kamailio.bindport is not used for the purpose here.
To change the listen port modify the parameter "Listen" or "port"
On Oct 3, 2015 3:01 AM, "amjad ali" wrote:
> Hi,
>
> I am scratching my head for one month to get this sorted and I have spent
> so many hours but