Are you configuring and using in script the aaa_flag (
http://www.opensips.org/html/docs/modules/1.8.x/acc.html#id292429) ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12/18/2012 04:09 PM, Willian Mazzardo - SYSSVOIP wrote:
I have made some adjusts in freeradius and radiusclient-ng files...
and my acc module on opensips.cfg is:
modparam("aaa_radius", "radius_config",
"/etc/radiusclient-ng/client.conf")
modparam("acc", "aaa_url",
"radius:/etc/radiusclient-ng/radiusclient.conf")
modparam("acc", "aaa_extra", "via=$hdr(Via[*]); email=$avp(s:email);
Bcontact=$ct / reply")
Need I put something in route script?
Thanks
Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br <http://www.syssvoip.com.br>
55 3537 2030
2012/12/18 Willian Mazzardo - SYSSVOIP <[email protected]
<mailto:[email protected]>>
Ok. I will do that.
Thanks
Em 18/12/2012 05:06, "Bogdan-Andrei Iancu" <[email protected]
<mailto:[email protected]>> escreveu:
Take a look at http://www.opensips.org/Resources/DocsTutRadius
And be sure first that OpenSIPS (properly configured) is
sending the ACC request to the RADIUS server.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12/18/2012 03:58 AM, Willian Mazzardo - SYSSVOIP wrote:
Yes... I follow the tutorial in CDR tool website.
There is any way to check if everything is ok?
Thanks
It might be a silly question, but have you configured the
accounting via radius backend ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12/17/2012 11:49 PM, Willian Mazzardo - SYSSVOIP wrote:
OK ... I have made some tests and now I`m able to use
Dialplan module on Opensips-cp ... and are working good.
Now i`m trying make work CDRTool on this scenario ... but no
luck ... cdrtool daemon is running, freeradius too ... but
no data on radacct201212 table on radius database.
How can I debug cdrtool to see what is going on?
Thanks
Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br <http://www.syssvoip.com.br>
55 3537 2030 <tel:55%203537%202030>
2012/12/17 Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>>
Hi Willian,
Assuming that route(3) is doing routing to register
subscribers and route(5) is doing routing to PSTN and
inside these routes you do the t_relay(), I would
suggest moving the setflag for accounting before
triggering those routes. The main idea is to have the
setflag done before the call is forwarded to whatever
destination.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12/17/2012 08:19 PM, Willian Mazzardo - SYSSVOIP wrote:
Hi Bogdan ... sorry for this ...
I've initiated some tests with Opensips ... and almost
everything is working ...
Now, i`m trying do a separate route for internal
accounts calls and PSTN calls.
I`ve this script on INVITE:
if (is_method("INVITE")) {
if(uri=~"^sip:[55910][0-9][0-9][0-9][0-9]@*") {
xlog("Willian: passou por aqui PONTO A PONTO");
route(3);
setflag(1); # do accounting
}else{
xlog("Willian: passou por aqui SAIDA");
rewritehostport("177.126.178.106:5060
<http://177.126.178.106:5060>");
route(5);
setflag(1); # do accounting
}
setflag(1); # do accounting
}
My internal accounts start with 55910XXXX and my PSTN
calls are Country Code + Region Code ... like for
Brazil = 555588889999 <tel:555588889999>
Is this INVITE section right?
Thanks.
Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br <http://www.syssvoip.com.br>
55 3537 2030 <tel:55%203537%202030>
2012/12/15 Bogdan-Andrei Iancu <[email protected]
<mailto:[email protected]>>
Hi,
This is a mailing list for opensips project, and we
do offer support and help for opensips. So either
you redirect your question to the right mailing
list, either you start using opensips
Regards,
Bogdan
Sent from Samsung Mobile
Willian Mazzardo - SYSSVOIP
<[email protected]
<mailto:[email protected]>> wrote:
Hi all..
I`m a very new user coming from Asterisk, and I
want to do some test with Kamailio billing / cdr my
calls.
I have installed CDRTool and Kamailio with a
working cfg who route any call to my SIP Provider.
But, when I do some call and hang up later... the
system doesn't create any log into radacct* tables.
I checked every configuration in
/etc/cdrtool/global.inc and seems to be OK.
I think maybe is an kamailio routing issue, like no
flag or something.
Can anyone help me with this?
Thanks in advice.
Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br <http://www.syssvoip.com.br>
55 3537 2030 <tel:55%203537%202030>
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