Little less...
In my issue I have some organizations in OCS (each organization have self
sip domain and self PSTN GW).
All calls from OCS go to OpenSIPS (through OCS Mediation). I don't use TEL
URI instead SIP URI, so OpenSIPS receive something like this:
us...@domain1.loc, us...@domain1.loc, us
2010/4/27 Cesar DOliveira
> Hi
> ..
>
http://dag.wieers.com/rpm/packages/perl-Frontier-RPC/
Before you ask things about how you can use this repository, check out this:
http://dag.wieers.com/rpm/FAQ.php#B
-Laszlo
___
Users mailing list
Users@
Hi
I appreciate your raply here is the full screen I am try to install the
opensisp version (opensips-1.6.2-0.x86_64.rpm) but it never get intalled do
have 1.4.5 installed and so I do have one more question how I can open the
OpenSIPS Control sorry I only two weeks on this so I learning any hel
I'm not sure what you're specific issue is or what module you are
using. However if I were to see this, I'd likely try to install the
module with CPAN. :)
-Brett
On Mon, Apr 26, 2010 at 6:07 PM, Cesar DOliveira wrote:
> Sorry I need help!
>
>
> ---> Missing Dependency: perl (Frontier : : RPC2) i
Sorry I need help!
---> Missing Dependency: perl (Frontier : : RPC2) is needed by package
opnsips-1.6.2-0.x86_64 (leurent)
can someone help me how to fix this or where i can fim RMP file to fix this
problem.
Thanks for any help
Buy the way I am use CentOS 5.4
Cesar
This functionality is not possibile with the the current
implementation of the rating engine and call control.
Adrian
On Apr 26, 2010, at 9:25 PM, Philipp Hoffmann wrote:
> Hi list,
>
> we want to route out costumers 0800 numbers. There the called party
> must
> pay.
>
> If the call comes fr
Here's a -W trace, I always get 2 unauthorised then 1 authorised OK.
There is a 30-second time span from start to end.
#
U 79.121.180.192:38670 -> 92.63.137.209:5060
REGISTER sip:voipexpress.co.uk SIP/2.0.
Via: SIP/2.0/UDP 79.121.180.192:38670;rport;branch=z9hG4bK03148.
Max-Forwards: 20.
To: .
Fr
Hi list,
we want to route out costumers 0800 numbers. There the called party must
pay.
If the call comes from fixed network (prefix: 49) it should be cost for
example 0.07 eur per minute.
If the call comes from cell (prefix: 49173, 49177, ...) it should be
cost for example 0.30 eur per minute
Prepaid history is updated when DebitBalance is called by Call control
module from OpenSIPS.
If this function is called when the call ends (for which you must
properly configure OpenSIPS and MediaProxy ) than you do not lose
money but rather your simply have duplicated radius records likely
Hi Anca,
Yeah I upgraded to the latest Polycom firmware 3.2.3 (see User Agent below).
It seems Polycom only accepts application/xpidf+xml and
text/xml+msrtc.pidf. If I send if "application/pidf+xml" it doesn't
work, even it I forward on a ""application/pidf+xml" NOTIFY from valid
UA like X-Lite.
Hey Guys,
Sorry for all the dumb questions lately, been trying to work out whats
going wrong.
I make use of the prepaid_history table in CDRTool to calculate the
daily usage for clients, and then email them a summary as well as their
remaining balance. What I've recently noticed when doing an
FYI
I tried to get (look) the calls in the modules CDRViewer and Dialog and not
is possible.
The following files have the default configuration and the MySQL tables is
according the script.
/var/www/opensips-cp/config/tools/system/cdrviewer/local.inc.php
/var/www/opensips-cp/config/tools/syste
IS working
Thanks
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Monday, April 26, 2010 5:06 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Problem with Opensips-Location tables-
Hi Bogdan,
I never even thought to try that, thanks for the pointer! When I was
looking at the docs I never even thought to try the reply code as the
first param, whoops!
Thanks for the help
On Mon, 2010-04-26 at 14:33 +0300, Bogdan-Andrei Iancu wrote:
> Hi Brad,
>
> for acc_db_request(), if
This example shows how you can use the latest features in OpenXCAP and
OpenSIPS to store, publish and subscribe to changes for user Icons
that can be displayed in a buddy list of a Presence enabled SIP User
Agent
http://www.openxcap.org/wiki/Icon
Regards,
Adrian
__
Hello,
There is a new release of OpenXCAP, version 1.2.0. OpenXCAP is now
packaged for four Linux distributions: Debian Stable, Debian Unstable,
Ubuntu Karmic and Ubuntu Lucid.
This is a major new version with new features:
org.openmobilealliance.xcap-directory, oma_status-icon,
org.openx
Hi Carmelo,
routing between different SIP domains in typically done via DNS
(resolving the DNS part of the RURI).
OpenSIPS supports this by default - if you check the default config file
that comes with OpenSIPS, you will find the section when calls targeting
other domains are routed - also ch
right!
Chris Maciejewski wrote:
> Ah I see. So if I use
>
> save('table1');
>
> in one part of my config file, and
>
> save('table2');
>
> in another part of the same config file, will opensips remove expired
> contacts from both 'table1' and 'table2' ?
>
> Regards,
> Chris
>
> On 26 April 2010 11
Yes, you must read the documentation that comes with MediaProxy to
correctly configure it.
Adrian
On Apr 26, 2010, at 2:26 PM, samoh wrote:
>
> Hi everybody,
>
> I tray to install Cdrtool, I installed freeradius, it works, I
> installed
> cdrtool and I have Rating engine running at 192.168.0
Hi,
First of all we would like to thank the whole opensips team. For testing our
client, we are mainly interested in the presence and rls topics and found
that a pretty good set of features is available right now, so thanks for
your job.
During our tests we encountered an issue running opensips 1
Julian Yap wrote:
> I guess Polycom phones only accept application/xpidf+xml.
>
>
>
Hi Julian,
I kept searching what "application/xpidf+xml" is and it was quite hard
to find since it seems to be something not used any more. I found in the
ietf draft preceding the PIDF RFC
(http://www.jdrose
Hi everybody,
I tray to install Cdrtool, I installed freeradius, it works, I installed
cdrtool and I have Rating engine running at 192.168.0.190:9024 but the files
of configuration of radius client and mediaproxy don't set correctely
because I have this message Error: mediaproxy certificate file
Hi,
OpenSIPS project has 3 papers with Amoocon 2010 - different topics
presented by different people (see more on
http://www.opensips.org/Events/Amoocon2010 ).
1. OpenSIPS 2.0 - a programmable framework for SIP (by Bogdan-Andrei Iancu)
http://www.amoocon.de/talks/99
2. Integrated SIP and
Hello,
I'd like to have Opensips and Asterisk on two different networks (and
different sip domains), and make each ua in Opensips domain reachable
from the ua in Asterisk domain and vice versa.
how should I configure Opensips (and Asterisk, if anybody knows)?
thanks for your support!
__
Hi Bogdan,
I am aware each opensips instance can only ping it's *own* contacts.
What I wanted to achieve is to avoid OpenSIPs1 pinging UA2, as shown
in my diagram:
http://wima.co.uk/2x_opensips.pdf
I have now solved my problem in the following way:
Opensips 1: modparam("usrloc", "db_url", "user
Ah I see. So if I use
save('table1');
in one part of my config file, and
save('table2');
in another part of the same config file, will opensips remove expired
contacts from both 'table1' and 'table2' ?
Regards,
Chris
On 26 April 2010 11:48, Bogdan-Andrei Iancu wrote:
> No need to hack the co
Hi Doug,
A simple solution will be to move all the branches into some AVPs that
you can consume (one by one) in a failure route.
to move branches try:
$var(i) = -1;
while ( $(branch(uri)[$var(i)])!=NULL) {
$avp(i:100) = $(branch(uri)[$var(i)]);
$var(i) = $var(i) -1;
}
I guess Polycom phones only accept application/xpidf+xml.
On Mon, Apr 26, 2010 at 1:18 AM, Anca Vamanu wrote:
> Hi Julian,
>
> Julian Yap wrote:
>>
>> Hmm, I have this working now using SIP SIMPLE standard (by testing
>> with X-Lite user agent)... Unfortunately this doesn't look to be
>> suppo
Hi Brad,
for acc_db_request(), if you put in the first param a reply code, you
will get the code in DB record ->
acc_log_request("503 Server Failure", "acc");
Regards,
Bogdan
Brad Bendy wrote:
> Hi List,
>
> Im trying to see if acc can save replies that are user generated, for
Hi Daniel,
You have to ways to handle this problem:
1) at original INVITE, when you know the NAT status of both caller and
callee, you can store a RR param about this (a cookie telling if caller
/ callee was natted). The RR header will show up (as Route hdr) in all
sequential requests (see the
Hi Julian,
Julian Yap wrote:
> Hmm, I have this working now using SIP SIMPLE standard (by testing
> with X-Lite user agent)... Unfortunately this doesn't look to be
> supported by Polycom phones. Is there a way to translate the PUBLISH
> so the NOTIFY will go out in an appropriate format dependi
Hmm, I have this working now using SIP SIMPLE standard (by testing
with X-Lite user agent)... Unfortunately this doesn't look to be
supported by Polycom phones. Is there a way to translate the PUBLISH
so the NOTIFY will go out in an appropriate format depending on the
UA?
eg. PUBLISH is generate
Hi Erick,
Erick Chinchilla Berrocal wrote:
>
> I have the following problem
>
> According the manual
>
> modparam("usrloc", "db_mode", 0)
>
> The modparam directive configures the corresponding module. The usrloc
> module in the line is responsible for the location service. In other
> words, whe
No need to hack the code - the usrloc module does not have any "table"
param as it automatically operates with the tables given by save() and
lookup().
So, the expired entries should be removed from the corresponding table
without any extra cfg.
Regards,
Bogdan
Julian Yap wrote:
> If there's
Hi Chris,
A quick one - with such a configuration, the NAT traversal will not work
(due the restriction on destination of the NAT pinhole) - so why pinging
them ?
I can make a small patch to you to set nathelper for pinging only if the
record socket is local, if not, no pinging - this may
Hi Dan,
Your descriptions point to a blocked fifo process. Blocking maybe
because of some internal locking (you see 99% cpu usage) or some I/O
(normal cpu usage).
So, do the followings:
1) do "opensipsctl fifo ps" to see the PID of the fifo process
2) make fifo to block
3) check if the fifo pr
Hi David,
Let's try and see what's the parent process of the zombie procs -> check
with ps and correlate (for the name) with "opensipsctl fifo ps"
I guess the parent of the zombies should be the "attendant proc" . BTW,
are you running with the "respawn" patch ?
Regards,
Bogdan
David Cunningh
Most of the configuration tasks are related to setting up components
outside CDRTool environment. Experience with configuring and running
Apache with PHP, MySQL, FreeRadius, MediaProxy and OpenSIPS are
required to successfully complete a CDRTool installation.
Setting up CDRTool is not
2010/4/26 samoh
> You don't help with your reply !
>
>
http://cdrtool.ag-projects.com/wiki/Install
"You will also need a valid client.conf, you can copy the one in
/etc/radiusclient-ng/radiusclient.conf and adapt it to suit your needs "
Install radiusclient-ng if you don't have itand follow
Hi Julian,
Putting inuse as basic status is not really legal - since the pidf RFC
says that you can have only open and close
(http://www.faqs.org/rfcs/rfc3863.html).
Regards,
--
Anca Vamanu
www.voice-system.ro
Julian Yap wrote:
> I'm trying to hack together my own presence server. I have P
You don't help with your reply !
2010/4/26 Adrian Georgescu [via OpenSIPS (Open SIP Server)] <
ml-node+4961513-394704167-513...@n2.nabble.com
>
>
> On Apr 26, 2010, at 10:07 AM, samoh wrote:
>
> >
> > Hi everybody,
> >
> > I'm traying to install CDRTool since 3 days but no success, if any
> > on
On Apr 26, 2010, at 10:07 AM, samoh wrote:
>
> Hi everybody,
>
> I'm traying to install CDRTool since 3 days but no success, if any
> one can
> help me ?.
> I installed Opensips 1.6 and loaded aaa_radius.so module, I follow thi
> installation guide but I don't find /etc/opensips/radius/client.c
Hi everybody,
I'm traying to install CDRTool since 3 days but no success, if any one can
help me ?.
I installed Opensips 1.6 and loaded aaa_radius.so module, I follow thi
installation guide but I don't find /etc/opensips/radius/client.conf it
doesn't exists !!???,
I found many problems to follo
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