Greetings all,
I have a relatively large domain list and when performing a "opensipsctl
domain show" I get the following error on the console and only 75% or so
of the domains show up:
ERROR:mi_fifo:mi_write_tree: failed to write - EOC does not fit in!
Is there a place I should allocate
Hello Bogdan,
Yes, Razvan identified issue where $fs wasn't set properly. I fixed
this on script level not sure if this clean solution.
https://paste.fedoraproject.org/paste/lIzEh5Q1vd4XePW6oUgC2g
any insight thank you.
volga629
On Thu, Aug 30, 2018 at 1:20 PM, Bogdan-Andrei Iancu
wrote:
Hi Bogdan
Yes, It's the same scenario and same message. The call flow is:
Asterisk Dials(port 5070) -> Opensips (port 5060) forward to Queue -> Calls
local user
I'm using standard Queue scenario:
server1
Thank you Vasilev !
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 08/22/2018 11:19 AM, vasilevalex wrote:
I created issue and sent possible patch
Hi Dragomir,
You still need to recompile the radius module by hand, after patching
(as in 2.2).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
http://opensips.org/training/OpenSIPS_Bootcamp_2018/
On 08/27/2018 03:23
Hi,
I see the INVITE is received and sent out on the same interface :
207.210.246.38:5060, so there is no interface exchange, so no reason for
a double RR.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2018
Hi Daniel,
Are you sure you configured a proper SIP URI as "message_queue" in the
flow description ? My impression is you have an empty string there - and
OpenSIPS is trying to put the call on the queue (as there is no agent),
but the SIP URI is not valid.
Regards,
Bogdan-Andrei Iancu
thank you I'll give that a try
On Thu, Aug 30, 2018, 3:18 AM vasilevalex,
wrote:
> In the same situation I used dialplan and dynamic routing modules like
> this:
>
> # Get ID of destination Asterisk server according to CustomerID
> dp_translate("1", "$var(cust_id)/$var(dst_srv)");
> if
In the same situation I used dialplan and dynamic routing modules like this:
# Get ID of destination Asterisk server according to CustomerID
dp_translate("1", "$var(cust_id)/$var(dst_srv)");
if ($var(dst_srv)==NULL) {
exit;
}
# Set route for SIP according ID of Asterisk server from Dynamic
