Hi,
I am getting this error in the log.
WARNING:core:get_send_socket: protocol/port mismatch
Any advice how to fix it?
Thanks
-Gary
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Thanks Bogdan & Liviu,
I have implemented the "failure_route" technique suggested by Liviu.
It is working fine.
Will try with t_relay("0x2") and let this forum know how it works.
I think it should fail and return -6 (- generic send failed), because the transport of the RURI (Callee Devi
Do you see the active calls number going up? Are you sure your calls are
hitting the block performing the set_dlg_profile. I'd put an xlog right
before the set_dlg_profile just to be sure. Depending on the db_mode you
might not actually see this value change in the DB. Either way, the
important thi
Hi Bogdan and the OpenSIPS community
I'm trying to limit concurrent calls on a group of sip devices, I got
stuck setting the profile in the dialog, its just not being set in the db.
I created to new tables, one that I store all the sip devices and to which
group they belong to, the second table
On 15 Jul 2014, at 20:51, Edwin wrote:
> Hi Saúl,
>
> In fact I have installed opensips and mediaproxy-dispatcher on Debian and
> the mediaproxy-relay on two other Debian servers. But tinyca2 had to run on
> a server with a Gui (like KDE or Gnome), which I (rather) don't run.
tinyca doesn’t ne
On 15 Jul 2014, at 22:55, Edwin wrote:
> Hi Adrian,
>
> I already did, but still don't get it (I know to google before asking). I
> really don't know what to do in what order (even after reading the readme
> and the openssl site a lot of times).
>
> I hope someone will just give an exampe (wh
Thank you Razvan. Great info.
Regards,
Ali Pey
On Wed, Jul 16, 2014 at 3:28 AM, Răzvan Crainea wrote:
> Hi, Ali!
>
> Rtpproxy offers an interface to communicate with it over network, through
> the communication socket (the -s parameter or the RTPProxy server). You can
> send over UDP a message
Hi Chris,
So the record is not found in memory. You can try to following - run the
MI command "subs_phtable_list" to list you all the subscriptions from
memory (watchers). Do that after the initial SUBSCRIBER (to see your new
subscription) and after that from time to time (like 20-30 secs) to
Hi,
The logs show your OpenSIPS trying to connect via TCP to the 80.x.x.x
IP, but not connection accepted. Is that destination supposed to accept
TCP conns ??
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 16.07.2014 14:27, mimic...@gmail.co
Gary,
Maybe take a look at the 0x02 flag for t_relay() :
http://www.opensips.org/html/docs/modules/1.12.x/tm.html#id294571
by default, t_relay() is internally sending a negative reply if not able
to send the request out (like DNS failure, bad IP, network error, etc).
Regards,
Bogdan-Andrei I
Or, be sure and set a reasonable dialog lifetime (like maybe 2 hours)
together with the "B" flag (to send BYEs one dialog timeout).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 16.07.2014 13:14, Schneur Rosenberg wrote:
create_dialog with P
My OpenSips version 1.11 has two domains (192.168.x.x and 80.x.x.x )
configured. and I can call devices within my LAN, however I am unable to
make calls from a device connected outside my LAN. The error message I
receive in SIP client (zoiper) says "bearer capability not authorized"
Captured from
Razvan,
Sorry just noticed your question on pseudovariable in BYE. Yes, you are
right, I did not set it in BYE but only in INVITE. In that case I accept
that there is no bug there.
DanB
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Hey Razvan,
I have hit the bug out of missing knowledge about how to gerate CDRs :).
For me there is no issue with BYEs since I only need INVITEs to generate
accounting.
Anyway, happy to contribute as bug finder ;).
Thanks again for your support!
DanB
__
create_dialog with Pp flag.
On Jul 16, 2014 10:45 AM, "Jorge Luis Ortea" wrote:
>
> Hi all,
>
> I'm using OpenSIPS 1.8. with several Asterisk. When Proxy SIP manages a
> call keeps a dialog with (called,caller,Asterisk) through store_dlg_value
> function from dialog module. Later through get_dial
Hi all,
I'm using OpenSIPS 1.8. with several Asterisk. When Proxy SIP manages a call
keeps a dialog with (called,caller,Asterisk) through store_dlg_value function
from dialog module. Later through get_dialog_info function it match calls and
resolves transfers issue.
I have the following p
Hello,
t_relay() will only "fail" because of internal reasons (e.g. out of mem,
max number of branches exceeded, transaction has 6XX status...).
It looks like you need to use a *failure_route* [1] for your
transaction, and perform your error handling when it eventually expires,
because no re
Hi, Dan!
You are right, this seems to be a bug. The second event shouldn't be
E_ACC_CDR, but E_ACC_EVENT. I fixed this in the git code, thanks for
noticing it.
Setting the CDR_FLAG on BYE messages means that you explicitly want to
account that message. That's why you have two events: a CDR and
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