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Hi Henk,
It doesn't work because even though b2bua is instructed to use the
socket on which it received the message to send the new message out, the
RURI does not point to a TCP contact. Setting a new ruri didn't work
until now ( I have just made now a commit in both 1.6 and trunk to take
Hi Takeshi,
No, it is not possible now to set a minimum expires value. If you need
that please open a feature request on out sourceforge project page.
Regards,
--
Anca Vamanu
www.voice-system.ro
On 11/29/2010 03:27 PM, mayamatakeshi wrote:
I can see there is a parameter
Hi Alex,
On 11/29/2010 05:15 PM, Alex Massover wrote:
Hello,
I'm trying to understand what is the status of OMA XDM in OpenSIPS. I
understand that even OpenXCAP supports now XDM to manage buddy list,
OpenSIPS is missing the module that supports it.
Is it correct? Is such module planned?
Hi Anton,
a.zagors...@oyster-telecom.ru wrote:
So, you explained how to store and grouping counters. Thanks.
But how to catch the moment when the opensips transaction is going
down? I need to react immediately.
when a dialog ends, opensips automatically removes it from the profile,
so the
Hi Brett,
As Anca said, look first if the branch in top most via is the same for
the INVITE you sent out and for the reply you received...
If you want, I can send you a small TM patch that will provide more info
on why a reply did not match a transaction
Regards,
Bogdan
Brett Nemeroff
Hi Mosbah,
you mean to use multiple interfaces with SCTP ? yes, you can
Regards,
Bogdan
mosbah abdelkader wrote:
Hello,
Does OpenSIPS supports SCTP multihoming?
___
Users
Hi Erik,
In opensips config file, you need to use brackets ( '{' and '}' ) to
contain blocks of instructions.
Regards,
Bogdan
erik pepermans wrote:
Hi,
See in my debug log file for each registration following error, but
registration succeeds (using opensips 1.6.2-notls) :
Hi Nawfel,
The problem is in one of the end points as for a 200 OK calls, the 200
reply and the ACK is end-2-end.
If you have a trace, maybe I can help you to see if there is a
signalling problem.
Regards,
Bogdan
Nawfel Oujdi wrote:
Hello!!
I m new in opensips and i m testing the load
Hi Dimitri,
If you use RTPproxy with opensips, RTPProxy has the capability to dump
the RTP (for a call) into local filesMaybe this can help you.
Regards,
Bogdan
DM wrote:
Hi,
Is there any awailable howto/solution for doing lawful interception
with using opensips?
Since it's 98% about
Here it is: http://www.opensips.org/Resources/Install#toc4
--
Anca Vamanu
www.voice-system.ro
On 12/02/2010 04:11 AM, Pradeep Patil wrote:
Dear All,
Anybody please help me for installing Open-SIP
OpenSIPS :)
on Ubuntu OS.
Any document/procedure please share
--
thanking you in advance,
Hi Bodgan,
Unfortunately it won't work for me, as they want to receive the stream
directly(and realtime) to their box, so one stream, mixed both the callee
and caller's audio.
2010/12/2 Bogdan-Andrei Iancu bog...@voice-system.ro
Hi Dimitri,
If you use RTPproxy with opensips, RTPProxy has the
I see.In this case you need to use a media server of some sort
Regards,
Bogdan
DM wrote:
Hi Bodgan,
Unfortunately it won't work for me, as they want to receive the
stream directly(and realtime) to their box, so one stream, mixed both
the callee and caller's audio.
2010/12/2
Hi Paul,
OpenSIPS it self cannot register with other servers (the UAC
capabilities are limited).
But you can easily do that with sipsak - you can use this utility to
register opensips's contact to another registrar server.
Regards,
Bogdan
Paul Wise wrote:
Hi all,
Is it possible for
Hi Bobby,
You can try to do something like:
1) do load_balance() to get the less loaded destination
2) use ratelimit to check the cps for it
3) if too many CPS - do again load_balance() - this will get the
next less loaded destination, skipping the one already used.
Regards,
Bogdan
Thanks, I'll try it out and let you know.
Regards,
Henk Hesselink
On 02-12-10 10:59, Anca Vamanu wrote:
Hi Henk,
It doesn't work because even though b2bua is instructed to use the
socket on which it received the message to send the new message out, the
RURI does not point to a TCP contact.
I guess it much depends on what's the exact task of opensips in your
platform
Regards,
Bogdan
Duane Larson wrote:
I would say Yes. Very possible.. It will involve a good deal of
reading, studying and testing, but i am 99% sure what you want to do
has been done before.
On Dec 1,
2010/12/2 Bogdan-Andrei Iancu bog...@voice-system.ro:
Hi Paul,
OpenSIPS it self cannot register with other servers (the UAC capabilities
are limited).
But you can easily do that with sipsak - you can use this utility to
register opensips's contact to another registrar server.
The sipsak
Bogdan,
Would it be possible to run sipak from the opensips.cfg script? I'm also
looking for something like this.
Kind regards,
Erik
-Oorspronkelijk bericht-
Van: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org]
Namens Bogdan-Andrei Iancu
Verzonden:
I developed my perl script using Net::SIP to do that (and also to play
audio message without asterisk).
s
Il 02/12/2010 15:24, Erik Dekkers ha scritto:
Bogdan,
Would it be possible to run sipak from the opensips.cfg script? I'm also
looking for something like this.
Kind regards,
Erik
Hi Stefano,
Are you willing to share that script with us?
Regards,
Erik
-Oorspronkelijk bericht-
Van: users-boun...@lists.opensips.org [mailto:users-boun...@lists.opensips.org]
Namens Stefano Pisani
Verzonden: donderdag 2 december 2010 15:29
Aan: users@lists.opensips.org
Onderwerp:
Hi all,
I just started playing around with opensip and I've already tested it with
single calls, the problem Im having is when there are multiple calls coming
in, we see a lot of INVITES but somehow the calls does not complete, I mean
I dial and nothing else happens.
Im only sending about 20 cps
Hello Vic.
Please, could you post the output of OpenSIPS log (Whether stdout or log
file), to confirm is there any problem.
Also, what is your test topology?
Regards.
Sergio G.
On Thu, Dec 2, 2010 at 12:06 PM, Vic Jolin victor.jo...@gmail.com wrote:
Hi all,
I just started playing around
Here is the m line from an INVITE/200 after the messages were modified by
use_media_proxy in each direction. The call happened to be mine and although
it did not stay up long enough for me to be 100% sure, I think the person on
the other end of the line was someone other than the person who
Hi,
On 12/02/2010 07:04 PM, Richard Revels wrote:
Here is the m line from an INVITE/200 after the messages were modified
by use_media_proxy in each direction. The call happened to be mine and
although it did not stay up long enough for me to be 100% sure, I think
the person on the other end of
This is interesting. On one relay the ports are two apart from each other
between the INVITE and 200 like I am used to seeing, but on the other relay the
ports are all over the map. However, I've done some test calls, and verified
they used the second relay, and the audio set up fine on all
While running a load test over the weekend, we ran into a segfault several
times that looks like it was happening around the same area. This is in
revision 7406.
It doesn't really feel there's anything meaningful or useful in the core
dump, but perhaps looking at the code path could help. We
On revision 7081 (opensips trunk), we've run into a couple of situations
where cores were generated from a segfault. They both look the same,
contents attached.
#0 backup () at codecs.c:104
104 int n = old-len;
(gdb) bt full
#0 backup () at codecs.c:104
l =
2010/12/2 Erik Dekkers erik.dekk...@wvds.nl:
Would it be possible to run sipak from the opensips.cfg script? I'm also
looking for something like this.
Why do you need it?? Just run sipsak as an external process generating
a REGISTER in behalf of your proxy.
--
Iñaki Baz Castillo
I'm embarrassed to say how long I've wished for this capability in
Opensips, and I never considered this possibility. Thanks Iñaki!
- Jeff
On 12/2/10 7:22 PM, Iñaki Baz Castillo i...@aliax.net wrote:
2010/12/2 Erik Dekkers erik.dekk...@wvds.nl:
Would it be possible to run sipak from the
Hi Bobby,
Your backtrace was very helpful: it looked exactly like one I got several
months ago! It is caused by an un-initialized field when tm clones into
shared memory. I uploaded patch under ID *3047314 *back in Sep. See the 1st
comment and the 1st patch file. The patch hasn't been accepted,
Excellent, thanks for the info and the fix!
I've applied this patch to a development machine and tested with the
signaling capture that got us into this scenario, and it seems to be fixed.
To the rest of the community, is there any way we can get this verified and
applied to trunk before the
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