Hi all,
OpenSIPS is going to be live on the FreeSWITCH "ClueCon Weekly"
conference tomorrow, February 21st @ 12 PM CST!
On this edition, the duty of representing the project falls on my
shoulders. The main topic of discussion will revolve around the latest
FreeSWITCH integration features tha
Hey
I had a problem when receiving simultaneous CANCEL from customer and 200 OK
from gateway.
Seems that the first CANCEL was rejected, but the second CANCEL was
accepted. This second CANCEL did NOT go to the gateway, just Opensips
received and replied with 200 OK.
This is the log of the first
Yup, it's included in 2.3.3. We'll look into the db_virtual issue asap.
Cheers,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 20.02.2018 18:23, Ben Newlin wrote:
My apologies. I tried to verify the change history for that file since
I am not on the newest version and I
My apologies. I tried to verify the change history for that file since I am not
on the newest version and I didn’t see any commits after the one I reported. I
must have been looking at it wrong. Was that commit included in 2.3.3?
I am currently testing the migration with 2.3.2 due to an incompat
Hi Ben,
I hope you're running off the release packages, because this 2.3 issue
has been reported & fixed here [1], a month ago. Otherwise, we'll have
to do some more digging.
Best regards,
[1]: https://github.com/OpenSIPS/opensips/commit/d448ab6e3599d
Liviu Chircu
OpenSIPS Developer
http://
Hi,
I am running into an issue migrating from 1.11 to 2.3. It appears the dr_is_gw
function is broken in 2.3.x.
When not using partitions, the function parameters are not being translated
properly before being passed to the internal function. With this command:
dr_is_gw("$avp(src_uri)", "2", "
Hi.
My softphone is registered with the following AOR:
-
AOR:: g1r2u3p4o5
Contact:: sip:g1r2u3p4o5@127.0.0.1:50353;transport=TLS;ob Q=
Expires:: 10
Callid:: 53e387dc-81fe-45f
Hi, Mirko!
Your solution works too, you can stick with it, it basically has the
same effect.
Best regards,
Răzvan
On 02/20/2018 02:55 PM, Mirko Csiky wrote:
Hi Razvan,
in the meantime (yesterday), the problem was solved by using :
force_send_socket(tcp:192.168.12.175);
before t_relay()
In my
Hi Razvan,
in the meantime (yesterday), the problem was solved by using :
force_send_socket(tcp:192.168.12.175);
before t_relay()
In my database (i have 3 rows in load_balancer table only) it is so:
sip\:192.168.15.92
(\ as escape character for ":" DBTEXT syntax)
so, i think, it doesnt provide the
Hi Bogdan,
Mostly yes - single HI Header with multiple hi-entries.
A general solution/advice that would work with both single and multiple HI
headers would be much appreciated.
Thanks,
Xaled
From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
Sent: Tuesday, February 20, 2018 1:1
Hi Xaled,
You mean if a single History header contains multiple hi-entry instances ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit 2018
http://www.opensips.org/events/Summit-2018Amsterdam
On 02/19/2018 02:39 PM, xaled wrote:
Hi, volga!
Can you send a SIP trace of this failed call to my mailbox? It's one of
those "it's working for me" situations. The mid_reg_lookup() should be
smart enough to locate a contact based on its unique ContactID, when the
call comes in from FS.
Regarding regid: it's completely gone now
Hi, Mirko!
If you provision a UDP URI in the database, then the outgoing INVITE
will be sent as UDP.
If you want to preserve the protocol, regardless the URI's protocol,
you'll have to explicitely hack it in the script by altering the
destination URI, something like:
if ($rP == "TCP")
$d
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