Hello,
The beta release for 1.10 got green light for 20th of March - still many
things to do, many new exciting features to added, many fixes to be done.
I really want to thanks to all the people contributing to this
tremendous work - providing patches, testing, reporting, fixing bug. It
is
Hello
I'm trying to send NOTIFY message via mi_xmlrpc.
Opensips version:
version: opensips 1.10.0-tls (x86_64/linux)
flags: STATS: Off, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE
sorry, I meant 1.11 beta, not 1.10 :)
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 07.03.2014 12:34, Bogdan-Andrei Iancu wrote:
Hello,
The beta release for 1.10 got green light for 20th of March - still
many things to do, many new exciting features
Hello all,
The poll ended - many thanks to all who expressed options - the result
is very important to use as a guidance in the relation with the
community. Speaking of results, here are the final ones:
http://www.opensips.org/Community/Polls-SummitResult
For Europe, definitely we
Great News !!
--- Jayesh
On Fri, Mar 7, 2014 at 4:09 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote:
sorry, I meant 1.11 beta, not 1.10 :)
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 07.03.2014 12:34, Bogdan-Andrei Iancu wrote:
Hello,
Hello Tony,
That's definitely a good idea that must be considered - of course, such
a change is not trivial, but I will open a TODO ticket on the GitHub
tracker. Or if you have a github account, it will be wonderful if you
could open the bug report.
Best regards,
Bogdan-Andrei Iancu
Hello!
I’ve got opensips-1.10.0-1.el6.x86_64 and work with radius-acct.
I’ve the next problem:
If caller ends the call - all works properly, and
radius_send_acct(acct_answer»); send Radius-Acct message. But if callee ends
the call - radius_send_acct(acct_answer») does nothing.
In log:
Hello all,
I would appreciate your input/opinions in the matter of deprecating the
mi_xmlrpc module in favor of mi_xmlrpc_ng + httpd modules.
Both modules offer the same functionality : XMLRPC backend for the
Management Interface (see ww.opensips.org/Documentation/Interface-MI-1-10).
The
Hi All,
I am facing a issue with my messaging on opensips. The scenario are as
follows,
1. When UserAgent have some device failure or network connectivity down in
that case user presence still maintained, I can see the disconnected user
still online using ./opensipctil online, in
Hi All,
I use Opensips 1.9.1 and have enabled RTP and Nating in the configuration,
Whenever I use to connect the calls using my 3G connection, call gets
connected but my voice is not being heard, whereas though wifi everything is
working fine. I tried connecting with Linphone I didn't face
Hello Alexander,
The log messages you posted translates into : when OpenSIPS handled the
BYE, the corresponding dialog was already terminated. As you are running
in full debug, could you upload on pastebin the entire set of logs
corresponding to that dialog ?
BTW, are you doing the
Your work firewall must be blocking packets when you test on 3G. The Wifi
must be within your work network !! I hope you are using RTPProxy or
MediaProxy to handle media when originated from NATed clients. If yes, you
dont need STUN and TURN as of now.
--- Jayesh
On Fri, Mar 7, 2014 at 5:01 PM,
Hi Jayesh,
Wifi network is not within my work network. I have disabled all the
firewall in my server. If firewall is blocking the packets, but I could
place the call using a another opensource app.
-Thanks
Rajesh
From: users-boun...@lists.opensips.org
Hi,
I want to install OpenSIPS on Linux .
This is my server -
Linux s099 2.6.18-128.el5 #1 SMP Wed Dec 17 11:41:38 EST 2008 x86_64 x86_64
x86_64 GNU/Linu
Is OpenSIPS compatible with it ?
Waiting for your response
Thanks and regards
Sanjana Bhutani
Here is an example CFG-file that works now:
The message 183 prefix and visible IP gateway. And that could be a threat
of fraud.
Here: if you use the function topology_hiding (); it does not happen a fair
exchange:
BYE comes to the message 404, Not here rather than 200 OK
I use client_nat_test to
Hello Everyone,
We are looking towards supporting inbound calling to our UACs, and would like to
know if it's possible to have OpenSIPS load userloc info in a REGISTER free
environment (ie, upon initial invite).
What would be some effective ways to configure OpenSIPS to manage this type
of
Hello,
Once again, to hide the network topology from the SIP signalling you
need to use the dialog module with topology hiding:
http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id296001
On the RTP side, use any of rtpproxy or mediaproxy.
Best regards,
Bogdan-Andrei Iancu
Hi,
Which SIP message is no longer matching the dialog ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06.03.2014 11:49, H Yavari wrote:
Hi Bogdan,
Can you give us some example?
I use opensips with rtpproxy. but when I use the
I do not see the log you mentioned :(..I see BYE was received and dialog
was properly terminated.
Do you do the radius stuff on the script section foe sequential
requests, right ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 07.03.2014
Hello,
OpenSIPS is designed to work on any UNIX system, so you Linux is ok.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 07.03.2014 06:41, Sanjana Bhutani wrote:
Hi,
I want to install OpenSIPS on Linux .
This is my server -
Linux s099
Hello Nick,
could you detail on what you mean by load userloc info in a REGISTER
free environment ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 07.03.2014 15:34, Nick Cameo wrote:
Hello Everyone,
We are looking towards supporting
Hello Bogdan,
Sorry for the lack of clarity. What we would like is the associated
tables (ie, dialog) to be loaded with the phones info such as IP
address, port etc.. on an INVITE rather than a REGISTER.
Our setup has other means of validating phones without having to
exchange credentials. Is it
Hi,
how to match a dialplan number length in 1.10 since in 1.6 there were
modparam(dialplan, match_len_col, column_name)
regards
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