Hi,
I am trying to use the xcap authorization for PA during the SUBSCRITPION message servicing.
I have the following code which work fine when auth parameter is set to 'none':
if(lookup_user("$tu.uid", "@ruri"))
{
if (!t_newtran())
sl_reply_error();
drop;
}
(handle_subscription("registrar"))
...
}
but when I switch to auth 'xcap' I cant see any query send to XCAP server simulated on apache according to the presence handbook.
Please point me what do I miss.
Thanks
Cheers
Tomasz
I using the example configuration file that comes with ser-2.0-rc1
No matter what I do, I cannot figure out what's causing this warning
/usr/local/ser/sbin/ser[43254]: Maxfwd module- initializing
/usr/local/ser/sbin/ser[43254]: WARNING: xl_mod_init: more IP
10.0.0.6 not used
I have two ip's assigned to the server, using one Nic card
Static IP 10.0.0.6
Virtual IP 10.0.0.4
NicCard em0
I have tried
--------------------
listen=10.0.0.4
fork=yes/no
---------------------
--------------------
listen-10.0.0.6
fork=yes/no
--------------------
--------------------
listen=10.0.0.4
listen=10.0.0.6
fork=yes/no
--------------------
--------------------
listen=em0
listen=10.0.0.4
fork=yes/no
--------------------
--------------------
listen=em0
fork=yes/no
--------------------
I always get the same error message.
Any ideas?
Hi,
my code has no connection to any BBUA. It is just providing the
maximum call duration information back to OpenSER which can later do
what u want it to do with this info.
I personally use yate (yate.null.ro) to provide call cut when the
calls are going out on PSTN, by adding an extra SIP Header in OpenSER
and then later process it in Yate to timeout the calls. I guess there
are unlimited possibilities later, if u have the information available
in OpenSER AVPs, including creating your own timer module which sends
a session cancelation order to RTPProxy (heard that it is possible to
kill the calls with RTPProxy, but never done it).
Let me know if I was clear enough.
Cheers,
DanB
On 7/2/07, A.M. <agentm(a)themactionfaction.com> wrote:
>
> On Jul 2, 2007, at 11:10 , Dan-Cristian Bogos wrote:
>
> > All,
> >
> > as a personal gratitude for all the efforts involved in opensource
> > projects, especially in OpenSER community, I would like to contribute
> > myself with a bit of code, investing in something I considered useful
> > for this community.
> >
> > Hereby, I inform u that the first release of FreeRADIUS-CDRTool
> > connector module is available for download and use at sourceforge.net.
> > Project link: http://sourceforge.net/projects/frad-cdrtool/
> >
> > Basically, FreeRADIUS-CDRTool is a FreeRADIUS module written in python
> > (therefore one might need to compile rlm_python in FreeRADIUS to make
> > it work), able to connect on CDRTool Engine via the Telnet socket (the
> > only way I found recommended by CDRTool folks) and perform different
> > actions via Prepaid API, like return maximum value for the call,
> > instruct CDRTool to lock accounts already in a call, instruct
> > unlocking at end of the call and balance debiting, etc. This module
> > should be used together with OpenSER radius authentication and
> > accounting.
> > If proper configured, FreeRADIUS-CDRTool can authorize calls for
> > CDRTool Prepaid users. At the starting of each call the module will
> > return the maximum call duration allowed, information which can be
> > processed later in a bbua scenario to timeout the calls reaching 0
> > balance, and total credit available at starting of the call, all
> > through SIP-AVP Radius Attributes.
>
> Hi- thanks for the information. How does your code interact with the
> b2bua? Which b2buas are supported?
>
> Cheers,
> M
>
All,
as a personal gratitude for all the efforts involved in opensource
projects, especially in OpenSER community, I would like to contribute
myself with a bit of code, investing in something I considered useful
for this community.
Hereby, I inform u that the first release of FreeRADIUS-CDRTool
connector module is available for download and use at sourceforge.net.
Project link: http://sourceforge.net/projects/frad-cdrtool/
Basically, FreeRADIUS-CDRTool is a FreeRADIUS module written in python
(therefore one might need to compile rlm_python in FreeRADIUS to make
it work), able to connect on CDRTool Engine via the Telnet socket (the
only way I found recommended by CDRTool folks) and perform different
actions via Prepaid API, like return maximum value for the call,
instruct CDRTool to lock accounts already in a call, instruct
unlocking at end of the call and balance debiting, etc. This module
should be used together with OpenSER radius authentication and
accounting.
If proper configured, FreeRADIUS-CDRTool can authorize calls for
CDRTool Prepaid users. At the starting of each call the module will
return the maximum call duration allowed, information which can be
processed later in a bbua scenario to timeout the calls reaching 0
balance, and total credit available at starting of the call, all
through SIP-AVP Radius Attributes.
I have tested the module in a Debian Linux environment, python2.4 and
FreeRADIUS 1.6, CDRTool version 5.1.4, on low call load.
I would not recommend yet using it in a production environment till we
get some stability feedback.
Have fun!
Kind regards,
DanB
Hi,
I have this common problem
Jul 2 16:03:25 localhost mediaproxy[5566]: error: uncaptured python
exception, closing channel <rtphandler.CommandHandler connected
xxx.xxx.xxx.xxx:46948 at -0x5f92e794> (exceptions.NameError:global name
'StopRecordSerializer' is not defined [asyncore.py|readwrite|94]
[asyncore.py|handle_read_event|391]
[/usr/local/mediaproxy/modules/rtphandler.py|handle_read|441]
[/usr/local/mediaproxy/modules/rtphandler.py|process|569])
Mediaproxy version is 1.8.2 it's running on debian 4.0 etch, witch
python 2.4 (also tried 2.5), i have python-mysqldb module installed,
also python-dev-all . Problem still exist.
When i'm using 1.7.2 version od mediaproxy, it seems to be ok.
I'm also using 1.8.2 mediaproxy version on debian sarge with python 2.3
where everything seems to be ok.
Any ideas? Broken debian packages?
Dzejms.
Hello everybody,
since this is my first post and I am kinda new to OpenSER, bare with
me ;)
I have been trying to manipulate invites for the last 2 days and cant
find the proper way to reach what is expected.
my goal:
send invites to specific PBX (Asterisk for example) with only the
desired extension and not the full username, this is based on a
avp_check if that UAC needs only extensions, some do, some don't ;)
Here is what I do:
First off all I extract the real username (in subscribers) and
extension from the R-URI and place it in "$avp(callee_user)" and
"$avp(callee_exten)"
#############
my solution No1 is "working"
avp_pushto("$ru/username", "$avp(callee_user)");
avp_copy("$avp(callee_exten)","$avp(callee_tmp)/gd");
if( subst('/^To:(.*)sip:[^@]*(@[a-zA-Z0-9.]+.*)$/To:\1sip:$avp(callee_tmp)\2/ig') ){};
-> callee_user is 012345678
-> callee_exten is 200
-> $ru is sip:012345678@xxx.xxx.xxx.xxx;user=phone
-> subst [ To: <sip:012345678200@xxx.xxx.xxx.xxx;user=phone> ]
with [To: <sip:200@xxx.xxx.xxx.xxx;user=phone> ]
Those 3 lines work "fine", the testing Asterisk PBX gets the call
correctly. The problem is that I should not change the "To:" part, it
will make problems with call forwards etc...
#############
my solution No.2 NOT working.
if (subst_uri('/^sip:([0-9]+)@(.*)$/sip:$avp(callee_tmp)@\2;/i')){$};
doesn't work because OpenSER of course cant find 200 in the userloc....
#############
Generally I would think that changing the INVITE uri itself like in
solution No.2 is the best way, couldn't find any proper way to make that
work though.
Does anyone have an idea how to realize this properly, or even if this
makes sense ;) ?
thx in advance,
Patrick.
Hey everyone,
I'm using OpenSER 1.1.1 SVN version.
I'd like to save the Authorization Digest Username as it appears in header
into a variable (avp) so
that later I can use it to set an accounting attribute in the radius.
>From the SIP Packet the information is given like this:
Authorization: Digest username="dsdsds", realm="1.2.3.4",
algorithm=MD5, uri="sip:1.2.3.4",
nonce="xxx"
Where I would like to grab the username part "dsdsds".
>From the docs I could only conclude that doing it would be something like
this:
avp_write("$hdr[Authorization]","s:12");
avp_subst("$avp(s:12)", "/.*username=(.*),/\1/");
I'm not even sure about that piece of code, if someone can help...
Thanks,
Lir.
Hi,
I am using the RLS module for RLS service and resource list subscriptions. Ive done everything according to the presence handbook.
When SER sends NOTIFY messages to the subscribing UAs every instances of the subscribed resources is in the pending state.
How are those subscriptions handled by ser?? Do I have to include presence_b2b molude so that those subscriptions in the pending state would be handled??
Thanks in advance
Cheers
Tomasz
Hello everyone,
I have following scenario:
+----------+
| VPN |
|+--------+|
|| MRCP ||
|+--------+|
| || |
|+--------+| +---------+
|| Proxy ====| Solaris |
|+--------+| +---------+
+----------+
I want both sip messages and rtp stream to go through proxy between mrcp
servers and solaris box.
Currently I am transferring sip messages using call forwarding (avpops
module) and trying to use rtpproxy for rtp stream forwarding. For
rtpproxy to work I have set up NAT on proxy, but still have issues.
This is the invite packet sent from proxy to MRCP server:
INVITE sip:UAS@10.1.1.100:5060 SIP/2.0
Record-Route: <sip:10.0.0.1;ftag=3d423348;lr=on> <=##
Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bKe359.5e010b95.0 <=##
Via: SIP/2.0/UDP
10.0.0.6:5070;branch=z9hG4bK-d8754z-5212be016a2d3c2a-1---d8754z-;rport=5070
Max-Forwards: 16
Contact: <sip:sender@10.0.0.6:5070>
To: <sip:synthesizer@10.0.0.1:5060;transport=UDP> <=##
From: <sip:sender@10.0.0.6:5070>;tag=3d423348
Call-ID: MTZkYWE2MGYxNTgyNmY0YTRmNzkwNjAyNmU5OGQ0ZGU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY
Content-Type: application/sdp
Content-Length: 282
v=0
o=- 1183369050 1183369050 IN IP4 10.0.0.6
s=-
t=0 0
m=application 35050 TCP/MRCPv2 <=##
c=IN IP4 10.0.0.1 <=##
a=connection:new
a=setup:active
a=cmid:1
a=resource:speechsynth
m=audio 53384 RTP/AVP 0
c=IN IP4 10.0.0.1 <=##
a=mid:1
a=recvonly
a=direction:active
a=nortpproxy:yes
Lines with "<=##" are the lines with which I think I have problem.
Because mrcp servers receive 'To' and 'Via' with address that is outside
vpn, they cannot reply. As a result I send x times invite and end with
408 timeout.
Also there is problem with line "m=application 35050 TCP/MRCPv2".
rtpproxy changes that port, but according to mrcp specification I need
it to be exactly 9. Even when changing PORT_MIN and PORT_MAX both to 9,
I sometimes get port 10.
Could you please advice me what to do? Am I trying in right direction?
Will I have to alter the sources to make it work like I need to?
Thanks for your answers
Martin
Hi, Olaf!
I'm not sure what functions that I can use to include inside ser.cfg. Could
you pls let me know where I can see all those functions like
sl_send_reply(), so that I can use them.
Thanks.
Best regards,
Roa Yu
-----Original Message-----
From: serusers-bounces(a)lists.iptel.org
[mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu
Sent: Friday, June 29, 2007 8:47 AM
To: Olaf Bergmann
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] How SER communicate to other SIP server
Hi, Olaf!
I'll have a look and try on it.
Thanks.
Best regards,
Roa Yu
-----Original Message-----
From: Olaf Bergmann [mailto:Olaf.Bergmann@freenet-ag.de]
Sent: Thursday, June 28, 2007 5:34 PM
To: roayu
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] How SER communicate to other SIP server
roayu wrote:
> Hi, Olaf!
>
> Ok, but which section that I need to concentrate on the most? I've search
> through the whole documentation and I can't find the configuration/setting
> on the ser.cfg file. Really appreciate on your help...
Look at the "Hello World ser.cfg"
<http://siprouter.onsip.org/doc/gettingstarted/ch06.html>. For
unknown SIP domains, route[1] does the DNS-based forwarding
mentioned before. You should give it a try and see how it works.
HTH,
Olaf
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