(top hiding);
In local route:
remove_hf(User-Agent);
append_hf(User-Agent: $var(ua)\r\n);
Regards,
Anca Vamanu
On Fri, Sep 14, 2012 at 12:08 PM, Jorge Henrique Pinho
jorge-h-pi...@ext.ptinovacao.pt wrote:
Hi Duane,
This does not work, because b2b will generate a new message with a
modified
Hi Jorge,
You could achieve what you want if you set the $du to the content of the
Route header before calling b2b_init_request.
$du = $hdr(Route);
Regards,
Anca
On Tue, Sep 11, 2012 at 2:20 PM, Jorge Henrique Pinho
jorge-h-pi...@ext.ptinovacao.pt wrote:
Hi all,
Do you consider that
explains this mechanism.
Regards,
Anca Vamanu
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Hi,
You get this behavior because of the way of the way top_hiding is
implemented.
The IP:PORT with which the contact is replaced corresponds to the interface
on which the message was received. It seems that in your case the message
is actually received on the private interface, so it will use
Hi Damien,
The behavior that you get is the correct one. If you publish with MI the
presence server will consider as if there is another device publishing for
the same account. So when sending Notify, it will aggregate what you have
sent with what it has received from the phone.
Regards,
Anca
Hi Damien,
So your idea is to have the possibility to decide from the script which is
the priority of a received Publish and to pass that to the presence module
as an extra parameter to handle_publish function? And this priorities to be
stored and used when aggregating the presence info.
Sound
Hi Damien,
On Sat, Oct 29, 2011 at 2:48 PM, Damien Sandras dsand...@seconix.comwrote:
Hello everyone,
When having multiple presence sources for a same SIP uri (e.g. different
devices or when mix_dialog_presence is enabled), OpenSIPS puts the newest
one on top. Consequently, most softphones
Hi,
Take the modules from 1.7 svn repo - this issue is fixed there.
Regards,
Anca
On Mon, Oct 10, 2011 at 6:12 PM, Jock McKechnie jock.mckech...@gmail.comwrote:
Greetings;
I'm trying to set up a, hopefully, very simple B2BUA top hiding
OpenSIPS 1.7.0 system... and I have no need for the
Hi,
You need to load dialog, pua and pua_dialoginfo modules:
http://www.opensips.org/Resources/PuaExtensions#pua_dialoginfo.
Regards,
Anca
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Hi Dani,
Does your phone actually send Publish for dialog event? I never saw this,
what phone are you using?
Anyhow, the Publish from the phone can not delete the information that
opensips has published with pua_dialoginfo because each record is identified
by a ETAG and when updating/inserting a
Hi Ryan,
You are right - the To and From headers in ACK are not the same as in
Invite in case the URIs contain parameters. I will handle this.
Thanks,
Anca
On Thu, Sep 8, 2011 at 4:40 PM, Ryan Revels r...@revelous.net wrote:
I'm running Revision 8257 from SVN branch of 1.6.
I've run into an
Hi Dani,
What do you mean by work with?
Opensips does send this reply in case the received etag doesn't match any
stored etag for that user.
Regards,
Anca
On Tue, Aug 23, 2011 at 6:19 PM, Dani Popa dani.p...@gmail.com wrote:
Hi,
should 412 Conditional Request Failed for PUBLISH (if
else Send INVITE to Media-Server to play some Message.
On Tue, Aug 23, 2011 at 12:22 AM, Anca Vamanu anca.vam...@gmail.comwrote:
Hi Sam,
Seems that you have a bad understanding of what the Refer scenario must
do. Let's say A and B are in a call. When a Refer message is received from
by sending first
an Invite without any SDP to B and doing a late SDP negotiation in 200OK and
ACK.
So what you observed, is in fact the wanted behavior.
Regards,
Anca Vamanu
On Mon, Aug 22, 2011 at 1:07 PM, Sam Govind govoi...@gmail.com wrote:
Hello,
I’ve been trying to configure the REFER
Hi James,
Try setting bla_fix_remote_target parameter to 0 :
http://www.opensips.org/html/docs/modules/1.6.x/presence.html#id248928
Regards,
Anca Vamanu
On Mon, Jul 25, 2011 at 12:52 AM, James Lamanna jlama...@gmail.com wrote:
Hi,
I'm trying to get BLF working on a Polycom 550 with Opensips
the callback
in b2b module and the b2b modules applies the body lumps when constructing
the pair 200 ok to send to the other end. Give it a try.
Regards,
Anca Vamanu
On Fri, Jul 22, 2011 at 12:31 PM, saneku uk...@ycc.ru wrote:
Hello!
I wonder is it possible to change sdp-fields on 200 OK reply message
Hi,
It is strange that setting $ru doesn't have effect. What version are you
using? Please try 1.7 if you haven't.
Regards,
Anca Vamanu
On Thu, Jul 21, 2011 at 9:33 AM, saneku uk...@ycc.ru wrote:
Hello!
if i remove $du=sip:##0950@10.130.0.137:5060 from cfg file asterisk
gives me 404 error
can check if there isn't like a race and a BYE is received
from Asterisk almost in the same time or before so that the b2b considers
the dialog ended. Watch closely the message trace.
Regards,
Anca Vamanu
On Wed, Jul 20, 2011 at 2:00 PM, saneku uk...@ycc.ru wrote:
Hello! thanks for you replay!
I
Hi again,
The 'load_balance' function sets the $du to the address of the asterisk box.
This wroks with b2b, the $du is used when routing the Invite out and since
you say the message does get to asterisk it proves that it works for you
also.
Since asterisk gives '404 Not Found' it means that the
))
{
rewritehostport(10.130.0.136:5060);
b2b_init_request(top hiding);
exit;
};
Regards,
Anca Vamanu
2011/7/18 Ухов Александр Ильич uk...@ycc.ru
Hello!
Please help! I am ready to broke my head . dont know how to solve this...
i have next: PSTN
(dialog vars, profiles, etc.).
You can read about how to use it in the module documentation page
http://www.opensips.org/html/docs/modules/devel/dialog.html#id294708.
The development for this feature was sponsored by Dynamic Packet,
dynamicpacket.com.
Regards,
Anca Vamanu
Hi Brett,
On Thu, Jun 30, 2011 at 5:40 PM, Brett Nemeroff br...@nemeroff.com wrote:
On Thu, Jun 30, 2011 at 9:12 AM, Anca Vamanu anca.vam...@gmail.comwrote:
A new feature has been added to OpenSIPS trunk and will be present in the
1.7 release: topology hiding functionality based on dialog
Hi Duane,
Sure looks like a Bria problem.. If it sends publishes with status closed,
the presence server doesn't have what else to do but to believe that is the
real state that it wants to publish. Maybe something in Bria's configuration
leads to this...
Regards,
Anca Vamanu
On Mon, Jun 20
Hi Denis,
We hit this problem also some time ago, it was indeed a bug when
applying lumps in local_route. We were just waiting for the fix to get
enough testing. It is stable now. I have just committed the fix in tm
module in both trunk and 1.6. Please upgrade and check.
Regards,
Anca
On
Hi Denis,
Sorry, I have to make a correction - I forgot that the fix that is
committed on svn is partial and works only if body lumps are applied. So
your case will still not work
and needs another fix. I suggest you to open a bug report in svn.
Regards,
Anca
On 06/14/2011 02:35 PM, Anca
#force_dialog
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On 05/13/2011 02:47 AM, Logan wrote:
Hello List. Supposedly the new trunk moves the vst/vsf variables from
the record-route header in to dialog. We're running the latest trunk
but the vst headers are still being written into record route. Am I
Hi Duane,
Well, this is exactly the purpose of the timeout_avp to terminate calls
after the period defined there. But, if you don't want the calls to
actually be ended after 2 hours, why don't you set it to a larger value?
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On 05/10/2011 08:00 PM
in on_reply route and for indialog requests in normal request
route. You could use subst function from textops module and replace
5000@ with 5001@ -
http://www.opensips.org/html/docs/modules/devel/textops.html#id250378.
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I am using opensips-cp
-bin/mailman/listinfo/users
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Hi Adam,
Please investigate the core file with gdb. Run 'gdb path_to executable
path_to_core' and then 'bt' and send the output.
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On 29/04/11 11:11, Adam Kuśmirek wrote:
Hi All
I need to implement Stateless Proxy to translate transport from TLS
Hi Paris,
Please read this: http://www.opensips.org/Resources/DocsTsMem.
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On 02/05/11 11:05, Paris Stamatopoulos wrote:
Hello everyone,
I've just noticed that my opensips 1.6.4.2 has been giving out an
error all day today
:
PKG_MEM_POOL_SIZE 1024*1024*4
( it sets 4M of private memory per process)
Regards,
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On 04/26/2011 05:44 PM, thrillerbee wrote:
Some more information... this only occurs when issuing 'opensipsctl
fifo dr_reload'. All other fifo commands seem to work without issue.
Anyone
is an endpoint.
Indeed there isn't a clear note in documentation about this limitation -
I will add it now.
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On 04/20/2011 06:26 PM, Anca Vamanu wrote:
Hi Dani,
Seems similar to something that we also hit.. but still not the same.
Can you please paste
for ERROR or CRITICAL messages
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On 04/20/2011 11:52 AM, Denis Putyato wrote:
Hello!
I found a big problem with Opensips in my installation.
Once I started receive acc record in acc table which had a big
duration (more than time conversation which
Hi John,
yes.. it is something in libxml library ( this is what presence uses). I
will try to find it.
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On 04/20/2011 06:51 PM, John Khvatov wrote:
Hi Anca,
On 20.04.2011, at 13:24, Anca Vamanu wrote:
Hi John,
Thanks for the logs.
Do you get
Hi John,
Thanks for the logs.
Do you get the memory statistics from opensips statistics or from top?
In the logs I don't see a shared memory leak..
Regards,
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On 04/20/2011 10:18 AM, John Khvatov wrote:
Hello Anca,
On 19.04.2011, at 13:21, Anca Vamanu wrote
the part of
the log with the dump to me ( not on the list, because it will be quite
large).
Thanks and regards,
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On 04/19/2011 10:30 AM, John Khvatov wrote:
Hello Anca,
Sorry, the memory leak is still here:
http://dev.sgu.ru/pub/opensips_memory_usage.2.png
Hi Jeff,
No, there isn't now a way to get in the script the received and
generated callid. But we could implement this, maybe by starting a list
of pseudovariables exported by the b2b modules and the first ones to be
- new callid, new from tag.
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anonymous due to
the public nature of this list and the fraud problems that we have
been experiencing.)
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functions
the way it has to.
So, I have two questions:
1) OpenSIPS B2B generated OK reply for both legs - is it normal? (dialog
module isn't used)
2) The reason of the ERROR message is the OK reply in a stage 4. Is this UA2
bug?
Thanks!
Regards,
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there the power of 2). The second int is the position in
the list for that hash_index.
Regards,
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Thanks
Pete
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Hi Duane,
For the pua, pua_dialoginfo and the presence part, I advise you to
update your code from svn. Recently there has been a change in them for
exactly this - cleaning up the records faster to avoid memory filling up.
Regards,
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On 04/06/2011 06:21
On 05/04/11 13:24, Anton Zagorskiy wrote:
Hi Bogdan,
Thank you, its working.
What about my first question - mixing two openSIPS on a same DB where second
openSIPS is just for b2b top hiding?
yes, it should be ok to use the same database.
Regards,
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WBR
-ID.
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On 01/04/11 14:51, Paris Stamatopoulos wrote:
Hi Anca,
Thanks for your interest on my issue.
Here are the NOTIFYs as being generated from the two OpenSIPS:
Initial Notify from Presence Service to Proxy OpenSIPS:
Regards,
Paris
*From
Hi Paris,
Hm.. that is strange, the contact is ok in Subscribe. The last thing
that I hope will clear things out - send the logs in presence server
when handling the Subscribe and generating the Notify. You can pastebin
them if they are too large.
Regards,
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header
fields in the CANCEL request MUST be identical to those in the
request being cancelled, including tags.
Regards,
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OpenSIPS Developer
It is less than ideal for us to contact their support and we'd like to get it
fixed asap. I've tried subst(), remove_hf and append_hf
Hi Paris,
Can you please also get a message trace with a Notify to see exactly
what happens with it?
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to implement end-to-end presence, then you have to forward
the Subscribe to the other client - just call lookup() and t_relay() for it.
Regards,
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OpenSIPS Developer
On 03/29/2011 06:02 PM, Stephen Bowman wrote:
Need some guidance on how presence should be configured.
Our setup
Hi David,
Have you configured OpenSIPS to check clients certificate (have you set
tls_require_client_certificate = 1) ? Then you have to configure the
accepted certificates:
http://www.opensips.org/html/docs/tutorials/tls-1.4.x.html#AEN264.
Regards,
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(x), ds is $var(ds)\n);
$var(x) = $var(x) + 1;
}
}
xlog(L_INFO,Destination set is $var(ds)\n);
}
Regards,
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OpenSIPS Developer
On 03/29/2011 01:00 AM, thrillerbee wrote:
I'm trying to get OpenSIPS to act as a REDIRECT server and have run
into a couple
Hi Dash,
Can you please raise the debug level to 6 and send a few lines above
those errors?
Regards,
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On 03/24/2011 12:07 PM, dashy dude wrote:
Hi Anca,
Thanks for the message.
I am using opensips-1.6.4-2-notls
I am able to see REGISTER with Expires=0
Hi Dash,
What OpenSIPS version are you using?
In the traces, do you see a Register with Expires=0 when the phone
unregisters? Also, look in opensips log- do you see any error?
Regards,
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On 03/23/2011 01:14 PM, dashy dude wrote:
Hi All,
I am quite new
Hi Iulian,
I suggest to update the dialog module from svn, branch 1.6. There was a
memory leak discovered after the 1.6.2 release and it was fixed in
January. It might also be showing up in your configuration.
Regards,
Anca Vamanu
OpenSIPS Developer
On 03/11/2011 11:39 AM, Iulian
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it to be preserved?
You are using the latest svn trunk? Ovidiu has implemented this
functionality - and the from dname is passed now.
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- Jeff
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you put DBENGINE=mysql at the beginning of the opensipsctl file?
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on that.. it's easy to add this. Put this
in the feature request also.
Anca
- Jeff
From: Anca Vamanu a...@opensips.org mailto:a...@opensips.org
Reply-To: OpenSIPS users mailling list users@lists.opensips.org
mailto:users@lists.opensips.org
Date: Wed, 23 Feb 2011 06:09:15 -0500
To: users
Hi Carlo,
On 02/17/2011 11:48 AM, Carlo Dimaggio wrote:
Il 16/02/11 18.19, Anca Vamanu ha scritto:
Hi Carlo,
Here are my ideas about this.
One solution would be to insert into usr_preferences table a record
with the URI where the call must be forwarded for the user (you can
put the username
handling Invites - do avp_db_load for the caller and if a
record is found forward it to that URI.
Regards,
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On 02/16/2011 06:20 PM, Carlo Dimaggio wrote:
Dear all,
I'm thinking about a design of an unified communication system based
on presence status.
I would
Hi Abid,
I suppose you configured your SIP client so that it sends Subscribe and
Publish with a Route header ( a preloaded route) and OpenSIPS does not
allow this. You should watch the network trace to see exactly what happens.
Regards,
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On 02/14/2011 11
and Notify requests to the end client.
Of course, this way is not scalable.
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not dies.
Are you using the newest version of rtpproxy?
Any ideas what I am doing wrong ?
Thank you.
-- Kamen
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It is in OpenSIPS also, probalby you missed it in readme -
http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#rtpproxy_stream2xxx
.
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should also report this
to the rtpproxy list.
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the reINVITE check!\n);
I suggest you to print the message buffer here just to be sure. Use
xlog($mb\n);
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if ( is_method(INVITE) ) {
append_hf(GW: REINVITE\r\n
and only if a REFER is received, it will do extra actions.
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Hi Kamen,
I have just fixed this bug and committed the patch on both trunk and 1.6
svn. Please update your code.
Regards,
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On 02/10/2011 04:44 PM, Kamen Petrov wrote:
I noticed my opensips daemon is not updated according to the trunk.
I just did a new
Hi Robin,
To compile only one module use:
make modules modules=modules/xyz
Regards,
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On 02/09/2011 04:27 AM, Robin Malhotra wrote:
Hi,
Please let me know how to compile a module. I am a beginner.
Say a module xyz
usr/src/opensips-1.6.2-tls/modules/xyz
you got with the two contacts - and probably it was
because you closed the client and did not unregister.
You can test this by looking that there is no contact, open up the
client and close it. Also run a message trace from the beginning to see
clearly what the client sends.
Regards,
--
Anca
file, in the main
route, for the initial Invite.
Also can I restore it to the original once I have changed ?
You can let opensips to automatically restore the original URI
http://www.opensips.org/html/docs/modules/devel/uac.html#id249092
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it is not registered.
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stored in $avp(s:222) is the integer
-1. It should not be such a nuisance to you, if it is 0 it is success,
if it has a lot of numbers it is error.
Regards,
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On 01/26/2011 05:12 PM, Dovid Bender wrote:
Bogdan,
I am not sure if I fully understand what you wrote
file and reply if you find any.
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Toyima,
I am sorry, I don't have experience in setting up conference systems, so
I can not make a recommendation.
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On 01/27/2011 10:21 AM, Toyima Dias wrote:
Anca,
What conference system would you recommend for me? Asterisk? SEMS? may
be some
it is not enough to handle the signaling, you also need
a conference bridge to distribute the media to more parties.
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that number is in fact -1.
Seems like www_authorize returns error.
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by a B2BUA and
not to a Proxy; but i've seen this link:
http://www.opensips.org/Resources/B2buaTutorial#toc15
You also need a media B2BUA for conferences, so you can not use only
OpenSIPS B2B.
How can i achieve these features in OpenSIPS? Should i take care of
the REFER messages?
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Hi Kyle,
I think also the only solution is in textops module - subs_body function:
http://www.opensips.org/html/docs/modules/1.6.x/textops.html#id292763
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On 01/22/2011 01:17 AM, Kyle Haefner wrote:
Hi All,
I need to change the number
Hi Toyima,
You can use t_replicate function:
http://www.opensips.org/html/docs/modules/devel/tm.html#treplicate.
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On 01/24/2011 02:35 PM, Toyima Dias wrote:
Hello,
I have a requeriment which indicates that all initiation calls
(INVITES
. But for the use case that you
described, recording the SIP messages on another server, t_replicate is
what you need.
Regards,
Anca
Thanks in advance for your help
2011/1/24 Anca Vamanu a...@opensips.org mailto:a...@opensips.org
Hi Toyima,
You can use t_replicate function:
http
,
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should I use?
Thanks!
You can use permissions module -
http://www.opensips.org/html/docs/modules/devel/permissions.html.
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the script is to insert
directly into database - in presentity table a correctly formatted body
with the desired information
(http://www.opensips.org/html/docs/modules/devel/avpops.html#id292802).
Regards,
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On 01/15/2011 08:25 PM, James Lamanna wrote:
So from
'select * from subscriber where username='1000' and check what the
domain column says. Is it the same as the realm inserted by the phone in
the 'Authentication' header?
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On 01/14/2011 04:54 PM, Gareth Blades wrote:
I dont have multidomain support loaded.
I
://www.opensips.org/Resources/DocsCoreVar#toc83.
It seems that there is no function to get a random values. Of course,
you could use perl code for that.
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Hi Chris,
You need to set the path to the configuration file.
You should edit file opensipsctlrc ( localed in etc/opensips/ ) and set
STARTOPTIONS= -f /etc/opensips/opensips.cfg ( replace with the path to
your configuration file).
Regards,
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Anca Vamanu
www.voice-system.ro
On 01/14
-telecom.ru
www.oyster-telecom.ru
-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
boun...@lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Wednesday, January 05, 2011 3:53 PM
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] b2b refer scenario
Hi Anton,
01/05
in the xml scenario file you will see that there
is a rule for REFER there - this is how it know that when a REFER is
received it has to do what it has to do.
Try this out and you will see ;) .
Regards,
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Anca Vamanu
www.voice-system.ro
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b2b_init_request(refer), then the B2BUA will do topology
hiding except for when a REFER is received, so in fact you just need to
call only once b2b_init_request with refer scenario.
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Anca Vamanu
www.voice-system.ro
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.
Allow: INVITE, CANCEL, ACK, BYE.
.
Here is the section in RFC that says the RURI must match the contact:
12.2.1.1 Generating the Request
If the route set is empty, the UAC MUST place the remote target URI
into the Request-URI.
Regards,
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Anca Vamanu
www.voice-system.ro
, how to use B2B REFER scenario from b2b tutorail? In this case
method_value is METHOD_REFER.
b2b_init_request must be called on the initial INVITE. I will mark
it in red in the tutorial.
Regards,
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Anca Vamanu
www.voice-system.ro
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
.
Regards,
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Anca Vamanu
www.voice-system.ro
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Hi Chris,
Do you see any errors in the log file containing 'msg_presentity_clean'
string?
Have you set the 'clean_period' parameter in presence module?
Regards,
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Anca Vamanu
www.voice-system.ro
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http
documents is not supported. But we should implement this as it is much
requested. Can you please make a feature request entry with this on
sourceforge?
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-- juha
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On 12/20/2010 04:13 PM, Juha Heinanen wrote:
Anca Vamanu writes:
Yes, now referencing resource-list documents in a rls-services
documents is not supported. But we should implement this as it is much
requested. Can you please make a feature request entry with this on
sourceforge?
anca,
i
Hi Duane,
I have found the bug there - it was a wrong definition of xcap-diff
event and I also noticed it in message-summary. I have fixed it now. I
suggest you to update the code again.
Thanks,
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Anca Vamanu
www.voice-system.ro
On 12/10/2010 11:02 PM, Duane Larson wrote:
Thanks
, [{password, passw0rd}]}
]},
Also the port you set in the opensips configuration is not right - the
default port for the component is 5347, you should not set the xmpp_port
parameter and let it use the default one.
Regards,
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Anca Vamanu
www.voice-system.ro
On 12/13/2010 02:49 PM
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