nd Developer
> http://www.opensips-solutions.com
>
> On 10/28/2017 04:13 PM, Adam Raszynski wrote:
>
> Hi
>
> Could you suggest some working configuration of the dispatcher module for
> the following setup:
>
> - Flag gateway as failed (even if it's still
Hi
Could you suggest some working configuration of the dispatcher module for
the following setup:
- Flag gateway as failed (even if it's still responding to OPTIONS) after
some number of consecutive call failures. That's the easy part
- Automatically un-flag gateway after some time, to allow
Hi All,
I'm looking for paid support in fixing my opensips config file
All interested OpenSIPS hackers are welcome:
https://www.freelancer.pl/projects/Software-Architecture-Linux/Fix-OpenSIPS-configuration-add-TCP.html
Hope that's good group, I've searched but didn't find better place to post
Hi all,
I would like to know is it possibble to check in OpenSIPS if user that is
destination of a call is registered and not busy (no other calls to same
username) without sending INVITE request?
I need this to create some kind of dynamic routing with busy detection
Any tips?
As suggested I enabled error logging via error_route to put in log full SIP
message bodies
Unfortunetly it doesn't seem to work with this type of error:
ERROR:core:parse_uri: unknown URI param list excedeed
For other parse errors I see message dumps, but not for this
So for some reason I'm
Hi
When I use dispatcher module and it gets 50x response from destination host
it automatically marks that destination as inactive for some time period
I only need to disable destinations when they don't respond for OPTIONS
probing
50x errors should not mark destinations as inactive
How to do
#id250429
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 02/20/2013 12:31 PM, Adam Raszynski wrote:
Hi
When I use dispatcher module and it gets 50x response from destination
host it automatically marks that destination as inactive for some time
Hi
Recently I have discovered increasing amount of the following errors in my
logs:
Feb 13 09:36:59 node1 /usr/sbin/opensips[10458]: ERROR:core:parse_uri:
unknown URI param list exceeded
I see this error is repeated many thousands of times in my log
Questions:
- What does this error mean?
-
Hi
Tried versions: 1.7.2 and 1.8
I have problem with OpenSIPS taking 100% CPU while trying to reconnect on
MySQL connection lost.
I make database backups every night and then for about 15 minutes database
is locked during backup procedure to prevent data corruption.
I understand that OpenSIPS
Hi
I have problem with some buggy hardware ATAs and routers, some of them mess
SIP requests by adding empty lines (\n or \r\n) between headers.
For example:
Via: SIP/2.0/UDP 80.1.1.1:5060;branch=z9hG4bKe89.28ac75a.0
To: sip:80.1.1.1
From: sip:username@80.1.1.1
Hi
I have exactly the same problem with dispatcher module and MySQL
after succesfoul ds_select_dst i get error:
ERROR:core:new_avp: invalid AVP name!
ERROR:core:add_avp: Failed to create new avp structure
resulting 503 response back to UAC
I'm using 1.8.0-notls version from debian apt
I use the following code on all my production OpenSIPS servers.
It's CPU friendly and avoids being spotted by bots searching for open-relay
VoIP servers.
route{
# put it at the very beginning of route section
if($ua=~friendly-scanner) {
xlog(L_ERROR, Auth error for $fU@$fd from
Hi
In default opensips.cfg there is following line:
if (!db_check_from()) {
send_reply(403, Forbidden Auth ID);
exit;
}
Beside that I authenticate all calls by using proxy_authorize function
The problem is that some buggy/cheap ATA's can't be configured to use user
in From field to be
Hi All,
Simple scenario:
- OpenSIPS as call router to SIP termination provider
- I have no control on remote gateways and can't generate early media there
Current situation:
- After dialing a number user hears silence until call is routed by my
termination provider, call routing to mobile
OK, I found that append_branch is the way to go
I create two branches, first for early media and second for the real call
Now the question is how to kill early media branch when second branch
sends 180 or 183 reply?
___
Users mailing list
Hi all
I'm looking for OpenSIPS expert to help me create config file for use with
SIPPY B2BUA
For anyone interested, here's link for project details:
http://www.freelancer.com/projects/Linux-VoIP/Help-fix-OpenSIPS-configuration-for.html
Please feel free to PM me or contact via freelancer portal
Hi All,
I would like to log all INVITE requests sent by users with wrong
authentication details
The problem is that I only need really failed auth INVITE attempts (with
wrong username/password), not all requests sent with no credentials (before
challenged by OpenSIPS).
Following script logs
17 matches
Mail list logo