Hello,
I tried to use apt.opensips.org, but I got the follwoing:
apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 049AD65B
Executing: gpg --ignore-time-conflict --no-options --no-default-keyring
--homedir /tmp/tmp.RMTmIE5VUn --no-auto-check-trustdb --trust-model always
--primary-keyring
Hi John,
Use $si to read/write the "send socket" associated with the RURI.
Now, the discussion deviated and touched several topics, but trying to
understand the "There were too many problems with sharing one location
table for two different sites" - besides the socket aspect, is there
Hi,
Can you elaborate on why is this so bad ? We migrated from $avps to
$acc_extra to make the scripting easier and cleaner - so a better
experience for the users.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Bootcamp 2017,
Hi Schneur,
There is no "explode" like in php - I guess you need to stick to
s.select for the moment (some major changes on handling arrays from
script are planned for the future releases). To have a clean script,
simply make a route[] dedicated to unpack a comma separated list into a
Hello Oliver,
In 2.2 the tracing is a bit different (as more complex) from 2.1. See
the trace_id parameter:
http://www.opensips.org/html/docs/modules/2.2.x/siptrace.html#idp154384
You should do like:
/*DB trace id*/
modparam("siptrace", "trace_id",
"[tid]
Hi Bogdan,
The problem could be from setting db_mode to 3. As you said earlier, this is
always required when using a shared location table.
My DB table would have two records - one where the socket is for server A and
the other with the socket for server B.
It is possible that when the
John,
I tested with 2.4 (dev branch) having:
modparam("usrloc", "db_mode", 2)
It might be from the DB mode 3.are you sure that the socket info you
have in DB is local to the OpenSIPS loading the record ? otherwise the
socket info may be discarded when the OpenSIPS is loading the
David,
Have you read this:
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip#Migratingfromchan_siptores_pjsip-Disablingres_pjsipandchan_pjsip
It looks like you can only have both active together if they are listening on
different ports. Do you know which port each
Hi John,
I just tried your logic and it seems to work. I had 2 phones registered
under super_t...@opensips.org and made a call to that account:
Jun 6 16:20:19 voip opensips_proxy[2584]: test - 1 branches found while
routing INVITE to sip:super_t...@opensips.org
Jun 6 16:20:19 voip
In the 2.3.x version opensips moved from $avp variables to $acc_extra. This
is very bad, really very bad :)
--
View this message in context:
http://opensips-open-sip-server.1449251.n2.nabble.com/acc-extra-fields-tp7607614.html
Sent from the OpenSIPS - Users mailing list archive at
Hello Razvan! With [1] i could make it working.Thank you. -- С уважением, Денис.Best regards, Denis 06.06.2017, 13:28, "Răzvan Crainea" :Hi, Denis!Are you sure that the two opensips instances are reachable? Can you execute the 'clusterer_list' command on both servers?Also,
Hi, Denis!
Are you sure that the two opensips instances are reachable? Can you
execute the 'clusterer_list' command on both servers?
Also, is the dialog replication working at all?
Finally, replicating both dialogs and profiles is not such a good idea.
I think replication only the dialog is
Hi,
I'm having some problems when using topology hiding. In my scenario an
INVITE comes in to the opensips (SBC) instance from another opensips
instance (Proxy). It is routed to a callee which eventually answers with a
200OK. The 200 OK is routed through the SBC to the Proxy which answers back
Hi, Igor!
What version of OpenSIPS are you using? Starting with 2.3 you can pass
arbitrary commands to the rtpengine module that are passed further to
the rtpengine daemon.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 06/02/2017 10:55 PM, Igor Olhovskiy
Did you check your rsyslog (/etc/rsyslogd.conf) configuration ?
Michele
On 05/06/2017 10:52, Kirill Galinurov wrote:
> Hi ALL.
> I try to store cdr in flat files.
> My config is
> loadmodule "acc.so"
> modparam("acc", "detect_direction", 0)
> modparam("acc", "extra_fields", "log: src_ip;
Hi, Royee!
Can you turn on debugging for this call and send the logs (in private if
privacy is an issue)?
My assumption is that the dialog is created before fix_nated_contact()
is called, perhaps due to a siptrace() call or something, and the wrong
contact ends up in the dialog. Or perhaps
Hi Michele,
You got it right.
I'll take a look about group module.
Thanks a lot
2017-06-06 11:00 GMT+02:00 Michele Pinassi :
> Hi Lorenzo,
>
> if i figure out correcty, you want a context separation based on network,
> right ? You can do it in some ways: for example,
Razvan,
Thanks for the help!
fix_nated_contact is called before create_dialog as you can also see from
the configuration file I added in the previous email.
When using the MI command I see:
callee_contact:: sip:USERNAME@PRIVATE_IP:PORT;transport=TCP
So it seems like the dialog is created with
Hi Lorenzo,
if i figure out correcty, you want a context separation based on
network, right ? You can do it in some ways: for example, you can use
group module
(http://www.opensips.org/html/docs/modules/devel/group.html) and From
URI to detect to which group caller is belonging to:
if
Moving this discussion to github:
https://github.com/OpenSIPS/opensips/issues/1136
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 06/05/2017 01:46 PM, Sasmita Panda wrote:
I was just checked this flag in opensips-1.6 . And its working fine as
for my expectation
Hi, Royee!
Can you make sure that fix_nated_contact() is called before create_dialog()?
Also, if you run through MI the dlg_list_ctx command, what is the
contact header stored in the dialog?
Best regards,
Răzvan
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 06/06/2017
21 matches
Mail list logo