Dear all,
I use the db_text module for the dispatcher and domain tables. For the
dispatcher table, it works successfully. However, for the domain table, the
OpenSIP server always fails to read the domain table on startup. Please help
to take a look what listed below. Thanks for any comment.
Best
Hi,
What happens after drop() is called from a branch_route?
In my case, I armed a failure route for the incoming INVITE in the request
route and therefore, I would expect that the failure route would be called
after drop() is called. However, I see the following error in the logs and
the followi
Hi Adam,
Take a look at http://www.opensips.org/Community/Business
On Sun, Oct 5, 2014 at 4:36 PM, Adam Raszynski
wrote:
> 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/Softw
Hi, James!
Unfortunately the old link for radiusclient-ng no loger works and the
PKGBUILD was not updated for Archlinux. You can get the
libradiusclient-ng library from SourceForge[1], compile and install it
manually.
[1] http://sourceforge.net/projects/radiusclient-ng.berlios/
Best regards
Already make a bid as Erewin.
05.10.2014 20:36 пользователь "Adam Raszynski"
написал:
> 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-con
Hi All,
Collecting CDRs via rabbitMQ.., I am not clear about it. Do I need to
raise the event E_ACC_CDR manually in the script ? or this should be done
by ACC module automatically?
If it is automatic how can I do that? I really appreciate if you share your
ACC and RabbitMQ settings.
Thanks.
Lev
Hi,
OpenSIPS routes a call to a SIP device (destination).
And (say) the device returns "486 BUSY" to OpenSIPS.
Now the call needs to be routed to the Voice-Mail server (another SIP device).
How to do it in opensips.cfg?
Any ideas?
Thanks.
Best Regards,
- Gary
__
Hi all,
I am getting this error "ERROR:tm:t_check: INVITE reply cannot be parsed"
when someone makes a call though our system.
I am not sure where it came from. I am just wondering if anyone know some
common problem when this has happens?
Alec Doran-Twyford
| Support Engineer for IVSTel |
| E-ma
Hi,
Is it possible to know (inside a failure_route[] block), which timeout caused
to land up in the failure_route?
# E.g. in the main route, I have:
if(is_method("INVITE")){
...
...
$T_fr_timeout=8;
$T_fr_inv_timeout=30;
t_on_failure("1");
t_relay();
exit;
}
# And in
Hi Bogdan ,
Thank you for replying. Here is the the REGISTER request and its
corresponding 200 OK.
REGISTER sip:com.mydomain.net SIP/2.0
Call-Id: yl44SiUAAA@xxx.x.x.x
CSeq: 2 REGISTER
From: ;tag=rm44SiUBAA
To:
Via: SIP/2.0/TLS xxx.x.x.x:;branch=z9hG4bK1411213658669;keep
Max-Forwards: 70
Cont
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