Thanks for the reply, that works :)
What about if I want to delete or overwrite an header ?
I need to change to To Header. If i use append_reply() will it overwrite ?
Cheers
Hi All,
I am facing issue while SIP REGISTER. Allow methods and Allow Event SIP
Headers are the mission of 401 Unauthorized and 200 OK on Successful SIP
Registered.
Please find the attached document in which I mention both working and
nonworking SIP Message.
It would be appreciable if anyone help figures out this issue on Kamailio
server.
Thank you.
--
Regards,
Suresh Talasaniya.
Contact : +91-9724264776
Skype : suresh.talsaniya
Hello
I am following up on the improper sip dialog that has been fixed by storing
contact details into a htable.
I am trying to delete the key at the end of the call, so the table doesnt
grow exponentially.
I've added a event_route[dialog:end] {} block, but I see it is processed
BEFORE the BYE is processed in the request_route(). Therefore, when I
delete the data in the htable, I cant relay this BYE correctly.
Is there another event (or technique) I could use to purge the htable ?
doing from a cron on the cmd line is not so nice, and can delete data still
needed for current calls
thanks
J.
Hey Guys,
I have faced the issue when I am trying to store answered the call to
PostgreSQL database using freeradius.
*Mon Jan 22 06:48:39 2018 : Info: [sql] Unsupported Acct-Status-Type = 15*
For more details, find here with attached radius.log.
# freeradius -v
freeradius: FreeRADIUS Version 2.2.5, for host x86_64-pc-linux-gnu, built
on Aug 10 2017 at 07:25:15
Copyright (C) 1999-2013 The FreeRADIUS server project and contributors.
There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A
PARTICULAR PURPOSE.
You may redistribute copies of FreeRADIUS under the terms of the
GNU General Public License.
For more information about these matters, see the file named COPYRIGHT.
Can someone help me to resolve this problem?
--
Regards,
Suresh Talasaniya.
Contact : +91-9724264776
Skype : suresh.talsaniya
Hello,
Kamailio SIP Server v5.1.1 stable release is out.
This is a maintenance release of the latest stable branch, 5.1, that
includes fixes since the release of v5.1.0. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.1.x. Deployments running previous v5.1.x
versions are strongly recommended to be upgraded to v5.1.1.
For more details about version 5.1.1 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2018/01/kamailio-v5-1-1-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
Tell me please.
Logs periodically display messages
2018-01-22T14: 14: 46.7701 "INFO: uac [replace.c: 392]: replace_uri (): Already called uac_replace for this dialog"
2018-01-22T14: 14: 46.7704 "INFO: uac [replace.c: 401]: replace_uri (): Deleted <_uac_funew> var in dialog"
The reason I know, the question is how in the dialog to overwrite the previously saved parameter, which was obtained with "uac_replace"?
--
С уважением,
Евгений Голей
Hello,
I am considering to do the first patch release from 5.1 series at the
beginning of next week (most likely on Monday or Tuesday), as usual, if
there are issues you are aware of and not reported to bug tracker, do it
as soon as possible in order to have a chance to be fixed:
- https://github.com/kamailio/kamailio/issues
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
Hello,
Sorry, I'm not that familiar with Kamailio. I am trying to figure out how to configure proxy-authorization for incoming sip invites.
I've defined WITH_AUTH in the configuration file but I think this only applies to sip register transactions.
Any insight is most appreciated.
Dave