Hello!
You can terminate an active call if you are using dialog support in your
script. This can be done using the dlg_end_dlg[1] MI command.
[1] http://www.opensips.org/html/docs/modules/1.7.x/dialog.html#id295086
Răzvan
On 03/17/2012 09:53 AM, goup2010 wrote:
Hello all,
How to terminate
Hi, Xi!
In order to use the MI fifo, you should first take a look at the module
documentation[1]. You have there an example[2] of how a request should
be built, and also the syntax grammar[3].
Just out of curiosity, why don't you use directly the opensipsctl tool?
[1]
Hi, Pieri!
You can get the desired behavior using dialog profiles. Take a look at[1].
[1] http://www.opensips.org/html/docs/modules/1.8.x/dialog.html#id249148
Regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 06/08/2012 03:29 PM, Pieri Mavarez wrote:
Hi, i
Hi, James!
I am not sure what exactly is the problem, but I have an assumption.
When the scenario was failing, where exactly was the $avp(jimmy) first
declared? Inside the LUA script? Is it now first declared in OpenSIPS?
Is there any chance you could reproduce this issue again in full
Hello, Arjun!
Are you sure rtpproxy is running and listening on that port? Have you
checked rtpproxy log for errors?
Regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 06/25/2012 05:58 PM, Arjun Shankar K S wrote:
Hi All,
Greeting to All!!
I have
20081102
Can you please let me know where is the mistake happening. Any help is
deeply appreciated.
Thanks,
Arjun
- Original Message -
From: Răzvan Crainea raz...@opensips.org
To: users@lists.opensips.org
Sent: Tuesday, June 26, 2012 1:21:11 PM GMT +05:30 Chennai, Kolkata,
Mumbai, New
Hello, Arjun!
There are many reasons OpenSIPS can generate a 500 Server Internal
Error, or perhaps you explicitly send if from script on some error
cases. The best thing to do in these situations is to check in OpenSIPS
logs.
Regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Duncan!
It seems like the Rasperry Pi cannot allocate more than 250 semaphores
at once, but the usrloc module and dialog try to allocate more than
that. In order to get fewer errors, you should configure OpenSIPS to
allocate for each module less than 250 semaphores. For the dialog and
Hi, Duncan!
It seems you have discovered two issues: an inconsistency (usrloc vs
dialog) of the hash_size parameter and a small bug in dialog module. For
the first one, the dialog module desires the exact number of the hash
size, but the usrloc module requires a power of two. Therefore, the
Hi, Sebastian!
The problem is that engage_rtp_proxy function does not support media
bridging functionality. In order to implement the scenario described
successfully, you will have to configure rtpproxy manually, using
rtpproxy_offer/rtpproxy_answer.
Best regards,
Razvan Crainea
OpenSIPS
Hello!
No, currently OpenSIPS doesn't support multiple notification
sockets.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 08/13/2012 01:25 PM, dpa wrote:
Hello!
You can use for several sets, but only a single socket for all
of them.
Regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 08/13/2012 03:56 PM, dpa wrote:
This message was generated by the Security Alerts service ( Free Trial 14th of
August - 14th of September )
http://www.opensips.org/Resources/AlertsMain
*
SVN commit*:
http://opensips.svn.sourceforge.net/viewvc/opensips?view=revisionrevision=9234
*Severity*: Medium
*Version* : 1.8, trunk
Hi, Flavio!
I am not sure that you should see that debug message. Have you tried to
take a trace on port 7891 and see if RTPProxy is sending any message?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 09/14/2012 08:05 PM, Flavio Goncalves wrote:
I
Hi, Nathaniel!
The lack of media is not the problem. It seems like one of the OpenSIPS
processes (pid 28602) doesn't initiate a proper communication channel
with RTPproxy, or some data was corrupted. Can you please send me
(privately if preferred) OpenSIPS logs so I can check this?
Best
access to. I was pretty large, so I thought that would be
better. I also copied the opensip config file that we are using to
that directory.
Thanks
Nathaniel
On 9/18/12 11:57 AM, Răzvan Crainea wrote:
Hi, Nathaniel!
The lack of media is not the problem. It seems like one of the
OpenSIPS
Hi, Julien!
First of all, welcome to the mailing list!
The dialog module can be configured to make all the dialogs persistent
in a SQL database, using the 'db_mode' parameter[1]. Apart that,
starting with version 1.8, the dialog profiles can also be used in a
distributed environment, by
Hi, Brett!
I think the problem is that the 'use_next_gw' function deletes the
gw_id_avp of the failed leg, and therefore the acc module cannot see
it. My suggestion is to use a different avp for the db_extra parameter,
and for each leg to copy the gw_id_avp into that avp.
Let me know if this
Hi, Mariana!
The dialog timeout avp has effect only if it is set before match_dialog
and loose_route. But only the dialog matched by either loose_route,
either match_dialog, will be updated. If none is found, then nothing
will be changed. Therefore I can't really see a problem here - if the
Hi, Ignacio!
The ports you have listed in the SDP snippet belong to a single rtp
stream - Callee-RTPProxy-Caller. You should also check the ports in the
200OK.
The nortpproxy_str parameter you are specifying is used by RTPProxy to
determine if the SDP has to be changed, or somebody else
192.168.1.220, I'm not sure if this is
correct.
If i don't put the domain parameter in the rtpproxy_offer the SDP
message contains the private ip of the RTPPROXY (192.168.1.220) and my
clients are outside this nat.
Thanks
2012/11/14 Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org
Hi, Jeff!
There were some fixes relate to the memory statistics in revision #9322.
Anyway, the problem was that the statistics were not reported properly,
but the internal behavior was working as it should. Therefore, your
OpenSIPS did run out of memory. Try to update your sources and then
Hi, Wesley!
I see that your modparam contains the localhost ip, however, the
OpenSIPS warning contains the ip_address. Is this ip_address
127.0.0.1? Because if it is not, then it means that the rtpproxy_sock
is set to a different value - either by overwriting the parameter, or by
using the
-solutions.com
On 11/28/2012 05:53 PM, Wesley Volcov wrote:
Dear Răzvan,
*
*
I tried to change the modparam configuration, to the ip_addres, and
also the rtpproxy listen addres to localhost, and it still the same
problem.
Regards,
Wesley Volcov
On 28 November 2012 12:31, Răzvan Crainea raz
temporarily
Nov 29 08:25:42 opensips /usr/local/sbin/opensips[5994]:
WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
Any idea?
Regards,
Wesley Volcov
On 28 November 2012 16:02, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org wrote
Hi, Wesley!
How exactly have you used the E/I flags? Have you tried to combine them
rtpproxy(ie) or rtpproxy(ei)? Taking a look in the Rtpproxy code,
it's kind of awkward how these flags are handled.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On
Hi, Flavio!
I've attached a patch for the rtpproxy module, that for every error, it
prints the buffer received from the Rtpproxy server. What I want to see
is if the message is totally broken, or only some parts of it are
malformed. Could you please apply this patch, and paste us the output
Hi, Brett!
The only problem I can see with your construction is that you do not
properly initialize the JSON object: $json(foo) = {}. When the =
operator is used, the {} is interpreted as a simple string, not as a
JSON format. You should have used the := which parses {} as a JSON
format and
#0122899.1149567571#0121
Flavio E. Goncalves
CEO - V.Office
2012/12/18 Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org
Hi, Flavio!
I've attached a patch for the rtpproxy module, that for every
error, it prints the buffer received from the Rtpproxy server.
What I want
and rtpproxy from the opensips
repository.
Flavio E. Goncalves
2013/1/7 Răzvan Crainea raz...@opensips.org mailto:raz...@opensips.org
Hi, Flavio!
Can you tell me what versions of OpenSIPS and RTPProxy are you using?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http
Hi, Chen-Che!
You can find the answers for your questions inline.
Regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 01/11/2013 04:33 AM, microx wrote:
Hi all,
I've encountered an issue as follows. Suppose two users communicate with
each other with one RTP
Hi, Chen-Che!
See my response inline.
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 01/11/2013 12:42 PM, microx wrote:
Hi Razvan,
Thanks for your reply. But can you explain a little more? Sorry for that I
cannot catch your idea.
From my experiment, when the RTP
Hi all!
We would like to announce you that the *OpenSIPS* project will attend
the 13th FOSDEM[1] conference on 2 3 February 2013, in Brussels,
Belgium[2], where I(Răzvan Crainea), Vlad Paiu and Saúl Ibarra Corretgé
will talk in the telephony devroom[3] about some of the most exciting
topics
Hello all!
As the OpenSIPS 1.9 release is coming up soon, we decided to release
some new enhancements for the Event Interface, that will facilitate the
interconnection with other applications and provide an easier way for
monitoring the OpenSIPS server. Following that, the latest version of
Hi, Brett!
Can you print in frame 1 the *pipe and **pipe.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 01/22/2013 11:38 PM, Brett Nemeroff wrote:
Hey All,
I'm getting a pretty regular segfault in 1.8.2 (svnrevision: 2:9596M)
Any idea what this
23, 2013 at 3:55 AM, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org wrote:
Hi, Brett!
Can you print in frame 1 the *pipe and **pipe.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 01/22/2013 11:38 PM, Brett
. The user has decreased the use of timeout notifications to a
minimum and the occurrence is now less frequent.
Still observing the problem.
Flavio E. Goncalves
2013/1/7 Răzvan Crainea raz...@opensips.org mailto:raz...@opensips.org
Hi, Flavio!
Can you tell me what versions of OpenSIPS
Hi, Brett!
We previously had a similar issue with the ratelimit module, but it has
been fixed in opensips 1.8. Are you using an older version?
Can you give a little more information about the ratelimit scenario?
What is the cachedb backend you are using for it? Is the limit shared
among
Hi, Dragomir!
Have you installed opensips 1.9.0 before executing the command? Or you
are still running it on the old opensips 1.8 installation?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/07/2013 08:37 AM, Dragomir Haralambiev wrote:
Hello,
Hi, Nick!
From what I see in your trace, the callee (Asterisk) is not sending
anything to RTPProxy. Have you tried taking a trace on the asterisk
ports to see if it is indeed sending anything?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On
Hi, Dragomir!
Your 1.9.0 installation has overwritten the 1.8 one? Or you've installed
it with a different prefix or something? Do you have the
nathelper-create.sql instead?
Can you execute ls -la /usr/local/share/opensips//mysql/ and check the
create time of the files?
Best regards,
Hi, Dragomir!
You should use the opensipsdbctl tool of the new installation, not the
1.8 one. That one should know how to locate the rtpproxy-create.sql file.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/07/2013 12:07 PM, Dragomir Haralambiev
Hi!
Can you please send me the output of the make install command? Perhaps
it would be better if you could send it directly to my email, as it is a
quite big output.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/07/2013 06:32 PM, Dragomir
menuconfig and enable the db_mysql module before the
installation in order to fix this.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/07/2013 06:35 PM, Răzvan Crainea wrote:
Hi!
Can you please send me the output of the make install command? Perhaps
Hi, Dragomir!
A fix was committed on both trunk and 1.9. Can you please update your
sources and install 1.9 again?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 02/08/2013 09:07 AM, Dragomir Haralambiev wrote:
Hi,
Thanks for your replay.
Here is
Hi, Seth!
Unfortunately OpenSIPS does not provide this kind of functionality.
Currently, the only way I can see this done is to execute an external
script that gathers the data.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/13/2013 12:00 AM,
Hi, Stas!
Currently the pseudo variables are not allowed to be used in the
modules' configuration. You can only use plain values.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/15/2013 03:22 PM, Stas Kobzar wrote:
Hello List,
Is there a way to
Hi, Jock
This doesn't seem to be an OpenSIPS issue, but rather a regular
expression one :). The 'replace' and 'substr' functions in OpenSIPS use
the POSIX regular expressions[1]. I am not an regexp master, and I don't
know how they work in perl either, so I can't conclude if you can
achieve
Hi, Khaled!
The match_dialog() function also matched the dialog and updates the
timer, therefore you should also set the pseudo variable for ACK before
the match_dialog() call.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/18/2013 03:38 PM,
Hi, Khaled!
Not really without an if close. You can use something like:
if (is_method(ACK))
$avp(timeout2) = 3540.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/18/2013 11:15 PM, M.Khaled W Chehab wrote:
Hi, Razvan .
You mean to add
Hi, Nick!
You said that you can see logs for RTPProxy. Can you set the debug level
to DBUG and paste (preferably on pastebin) the logs of the session?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/19/2013 03:52 PM, Nick Khamis wrote:
I wanted
Hi, Nick!
From your traces, I can see that the RTPProxy session is properly
established (you have both an offer and an answer). But on the media
level, all I can see is that Asterisk (the callee) is sending RTP to
caller, but the caller doesn't send anything. Also, this is what
RTPProxy
is coming from a different
source that of the SIP messages. So I think it's a matter of lining up
rtpproxy_offer/answer parameters (i.e., co).
Unfortunately, their service to our zone today is down. Will post
detailed logs as soon as we can initiate some calls.
Nick.
On 3/19/13, Răzvan Crainea raz
Khamis wrote:
I am using version 1.8.2 I think? I can check when I get back in the
office. I thought we had downloaded the latest stable not too long ago
but guess not. Is updating easy (i.e., no database wipe out?).
Nick.
On 3/22/13, Răzvan Crainea raz...@opensips.org wrote:
Hi, Nick!
What
Hi, Ovidiu!
Can you post (pastebin maybe) the entire log of the 'make deb' command?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 03/22/2013 10:31 PM, Ovidiu Sas wrote:
I got the similar errors on a freshly installed debian while running 'make
Hi Cindy!
for TCP: the error seems to occur while reading the message. Are you
sure the error route is not called for this type?
for UDP: Are you passing any flags to the sipmsg_validate() function?
Have you tried using the 's' flag [1]?
[1]
Hello!
It seems to be a problem with some Makefile variables that were not
overwitten. I have committed a fix on trunk, 1.9 and 1.8. Can you please
test again and let me know if there are any problems?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
in syslog.
I've also tried s, sh, and nothing as flags for sipmsg_validate.
Same error every time.
Thanks.
Cindy
On Mar 25, 2013, at 1:26 PM, Răzvan Crainea wrote:
Hi Cindy!
for TCP: the error seems to occur while reading the message. Are you
sure the error route is not called
Hi, Mariana!
Please upload the files on one of the online file storage services (like
[1]), and paste the links here.
[1] http://pastebin.com/
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 04/22/2013 01:58 PM, Mariana Arduini wrote:
Hello all!
.
Is there any email I can sent it privately, or could you give us any
guidance on how to analyse that info and get to any conclusion?
Thanks and sorry for the huge dump...
Mariana.
On Mon, Apr 22, 2013 at 9:08 AM, Răzvan Crainea raz...@opensips.org
mailto:raz...@opensips.org wrote
Hi, Pavel!
Are you using dialog-based accounting? Are you modifying the dialog timeout?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 04/23/2013 09:53 AM, pa...@eremina.net wrote:
Hi!
I've have 1.9 Opensips and get some confuse...
It can't save
Developer
http://www.opensips-solutions.com
On 04/23/2013 10:58 AM, pa...@eremina.net wrote:
Yes, i use CDR flag to account.
No, i don't touch any timeout value;
2013/4/23 Răzvan Crainea raz...@opensips.org mailto:raz...@opensips.org
Hi, Pavel!
Are you using dialog-based accounting? Are you
-solutions.com
On 04/23/2013 11:26 AM, pa...@eremina.net wrote:
I don't use flag B in create_dialog() because when i use it call may
drop with 90 second.
I understand that something wrong with timeout.
I use default dialog parameters if i understood you.
2013/4/23 Răzvan Crainea raz...@opensips.org
Hi, Brett!
Unfortunately there is no way to disable pinging. Create dialog can only
be called on the initial Invite, therefore it will fail for any
sequential requests.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/02/2013 07:27 PM, Brett
Hi, Mickael!
remove_hf() removes all headers, so it should remove both first and the
second one. Then you can simply add a new header(using append_hf()
function[1]) with the desired information.
[1] http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#id249181
Best regards,
will start migrating the
OpenSIPS sources to a new GitHub repository.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 04/30/2013 11:38 AM, Răzvan Crainea wrote:
Hi all!
Following the community desires[1] to migrate OpenSIPS repository and
tracker
Hi, Michele!
I noticed that you are using the engage_rtp_proxy() function, but it's
behavior is undefined when using RTPProxy in bridge mode[1]. Therefore
you should use manualy engage RTPProxy, using the rtpproxy_offer()
rtpproxy_answer() functions and the 'I' and 'E' flags to determine the
Hi, Qasim!
There are two problems with your approach: the first one is that you are
using the engage_rtp_proxy() function in a bridging mode scenario. The
behavior of this is undefined, because the rtpproxy module cannot fully
determine your scenario (for example what's the direction of the
As also detailed in the other ticket, as well as in the documentation,
the engage_rtp_proxy() function has an undefined behavior when using in
a bridged scenario. Therefore I recommend you to use the
rtpproxy_offer/answer() functions.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
() but the same problem. I will try using
rtpproxy_offer/answer() again in a bit more detail now specially after
hearing about problems in engage_rtpproxy in brigding mode. Now can you
point me how i can achieve nat handling in rtpproxy module?
Regards,
Qasim
On Thu, May 9, 2013 at 5:39 PM, Răzvan Crainea raz
with server's
local IP but on the other way arround it tries to send the IP back to
client's local IP address which is not visible to server.
Actually we have two nated acenerios. One on the server end and the
other on the client's end.
Regards,
Qasim
On Thu, May 9, 2013 at 5:59 PM, Răzvan Crainea raz
://www.opensips.org/About/GitHub-Migration
[2] http://sourceforge.net/p/opensips/_list/tickets
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/20/2013 12:45 PM, Muhammad Shahzad wrote:
OpenSIPS repo is frozen since May 08, 2013. Meanwhile i did couple
://www.opensips.org/About/Git-Migration
[4] http://git-scm.com/docs/gittutorial
[5] http://sourceforge.net/p/opensips/_list/tickets
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/08/2013 09:00 PM, Răzvan Crainea wrote:
Hello, all!
As per our initial schedule[2
Hi, Bobby!
I'm glad you're interested in enhancing OpenSIPS with extra features for
the Event Interface, and I am eager to help you with this.
The pvar interface in OpenSIPS is quite simple. Each variable consists
of a spec and a value, which can be string or integer. Reading or
writing a
Hi, Dani!
According to the RTPProxy protocol[1], you can find sessions information
using the 'I' and 'Q' commands. However, I am not sure they will be able
to provide all the information you require.
[1] http://www.rtpproxy.org/wiki/RTPproxy/Protocol
Best regards,
Razvan Crainea
OpenSIPS
Hi, Dani!
What you can do is to send a 183 from OpenSIPS without SDP (because you
can't possibly know the UAS media information), and inject the ringback
tone to UAC from RTPProxy.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/24/2013 07:39
Hi, Qasim!
The s and w flag do the exact same thing - instruct RTPProxy that
symmetric NAT should be used. You can use either of them and you will
get the same behavior. It is not similar to the i/e flags.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Sheetal!
Can you increase the debugging level for RTPProxy and send those logs
(preferably in a pastebin snippet)?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/27/2013 08:48 AM, Sheetal K Agarwal wrote:
Hi All
I am trying to record
Hi, Sunny!
It seems like the opensipsctl tool cannot find the mysql application.
Can you run 'which mysql' and see if you have it in your PATH?
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/28/2013 03:20 PM, Sunny Khan wrote:
Hello,
Please
Hi, Martin!
You are looking for the acc module[1]. In order to get CDRs in the
database, you should set the db_flag and cdr_flag for the initial INVITEs.
[1] http://www.opensips.org/html/docs/modules/1.9.x/acc.html
Best regards,
Razvan Crainea
OpenSIPS Core Developer
Hi, Jock!
What you should do is to catch the reply in an onreply_route and call
the append_hf() function for the 200 OK reply.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/28/2013 11:21 PM, Jock McKechnie wrote:
I'm sure this isn't this hard,
. Simply assinging a different
name (the first parameter of rl_check function), will result in creating
a new pipe (at runtime).
I hope my explanations facilitate your migration.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 05/30/2013 08:23 PM, Ximena
Hello all!
After migrating all the OpenSIPS repositories from SourceForge to
GitHub[1], the next step is to proceed with the tracker migration. We
wrote down two options here:
1. Move all tickets (patches, feature requests, bugs) to the GitHub
tracker. There are several problems with this
Hi all,
One more major release is out - OpenSIPS 1.10.0 (beta)
*OpenSIPS 1.10.0* comes with several improvements (asynchronous TCP,
better lumps management), but also with new functionalities like SCA
support with dialog module, a new Binary Interface used to efficiently
communicate with
Hello all!
As some of you might have already noticed, the old SVN repository
(hosted by SourceForge) does not contain the 1.10 branch yet. The reason
for not creating it yet is because we would like to argue a bit about
it's necessity.
The main reason we are still maintaining the old SVN
Hi, Miha!
You cannot use pseudo-variables in the seturi() function. However, you
can use the $ru pseudo-variable to construct the R-URI.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 08/20/2013 10:04 AM, Miha wrote:
Hi,
how can I use pseude
:
...
append_branch();
$rU = C;
...
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 08/20/2013 12:16 PM, Miha wrote:
HI Răzvan,
I am trying to implement this:
http://www.opensips.org/Documentation/Script-CoreFunctions#toc2
2. append_branch()
So if I do $ru=C instead
, but just first number from
array.
thx!
miha
Dne 8/20/2013 1:25 PM, piše Miha:
Hi Răzvan,
that I did:)
thank you for your help!
miha
Dne 8/20/2013 12:35 PM, piše Răzvan Crainea:
Hi, Miha!
According to the documentation, the second branch is created when
calling append_branch(), so yes
Hi, Marco!
If I remember correctly, there was a bug in OpenSIPS 1.7 related to
this, where the second parameter was always ignored. You should try to
update your sources to the latest version. Or, even better, upgrade your
OpenSIPS 1.8 or higher, since the 1.7 version is no longer supported.
in
production environments.
Thank you all for your help and contributions! Keep up the good work!
Cheers,
--
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
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Hi, Vladimir!
It seems like the Operating System is choosing an outbound socket that
is not bound by OpenSIPS. Are there any other interfaces that machine
has and you are not specifying them as listeners?
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
Hi, Luis!
Have you configured OpenSIPS to use mysql in the menuconfig tool? This
should be done before installing it.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 10/30/2013 11:08 PM, Luis Pérez Urteaga wrote:
Wilmar,
This is my list of modules
Hi, Aldo!
Can you please pastebin the coredump? You can find more info here [1].
[1] http://www.opensips.org/Documentation/TroubleShooting-Crash
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/05/2013 05:52 PM, Aldo Jose Spanghero Romao wrote:
Hi
Hi, Aldo!
How did you install OpenSIPS? Using DEB files, from sources? If you used
the first choice, have you also installed the debug package?
How did you narrow down the crash was in the rtpproxy_offer() function?
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips
Hi, Alexander!
Are you fetching the value before the loose_route() function call? It
should be available only after loose_route() or match_dialog() are
executed on the BYE request.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/08/2013 05:37 AM
Hi, Aldo!
Can you paste the line that generated the crash? It should be printed
right after gdb opens the core file, before you execute 'bt full'.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/07/2013 06:50 PM, Aldo Jose Spanghero Romao wrote
(BYE); from a different part of script.
You can't really extract internal dialog values, unless you explicitely
save them as values, or take them from the request.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/08/2013 11:17 AM, Alexander Mustafin
that calls match_dialog()) with the
dialog status, method and callid and match them with the logs in the BYE
route?
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/11/2013 09:24 AM, Alexander Mustafin wrote:
Hello!
I’ve tried many variants
Hi, Aldo!
I need the command that generates the core dump. You should see it above
the line you pasted and should be a C instruction.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/11/2013 06:33 PM, Aldo Jose Spanghero Romao wrote:
Hi, Răzvan
to eliminate the deprecated warning:
...
$avp(my_port) = 5060;
if (ds_is_in_list($avp(my_ip),$avp(my_port),1,1)) {
...
Best regards,
Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 11/20/2013 03:03 PM, Samuel Muller wrote:
Hey,
I would like to be sure that since
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