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<tt>Are you configuring and using in script the aaa_flag (
<a class="moz-txt-link-freetext" href="http://www.opensips.org/html/docs/modules/1.8.x/acc.html#id292429">http://www.opensips.org/html/docs/modules/1.8.x/acc.html#id292429</a>)
?<br>
<br>
Regards,</tt><br>
<pre class="moz-signature" cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a></pre>
<br>
On 12/18/2012 04:09 PM, Willian Mazzardo - SYSSVOIP wrote:
<blockquote
cite="mid:CAL2ykgEvgfGYdd2sSTUyTXbpDkOj3AEFTn_VqD6gEJPfpEWhuA@mail.gmail.com"
type="cite">I have made some adjusts in freeradius and
radiusclient-ng files... and my acc module on opensips.cfg is:
<div><br>
</div>
<div>
<div>modparam("aaa_radius", "radius_config",
"/etc/radiusclient-ng/client.conf")</div>
<div>modparam("acc", "aaa_url",
"radius:/etc/radiusclient-ng/radiusclient.conf")</div>
<div>modparam("acc", "aaa_extra", "via=$hdr(Via[*]);
email=$avp(s:email); Bcontact=$ct / reply")</div>
</div>
<div><br>
</div>
<div>Need I put something in route script?</div>
<div><br>
</div>
<div><br>
</div>
<div>Thanks</div>
<div class="gmail_extra"><br clear="all">
<div>Willian Mazzardo<br>
Depto TI - SYSSVOIP<br>
<a moz-do-not-send="true" href="http://www.syssvoip.com.br">www.syssvoip.com.br</a><br>
55 3537 2030</div>
<br>
<br>
<br>
<div class="gmail_quote">2012/12/18 Willian Mazzardo - SYSSVOIP
<span dir="ltr"><<a moz-do-not-send="true"
href="mailto:willian@syssvoip.com.br" target="_blank">willian@syssvoip.com.br</a>></span><br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<p dir="ltr">Ok. I will do that. </p>
<p dir="ltr">Thanks</p>
<div class="gmail_quote">Em 18/12/2012 05:06, "Bogdan-Andrei
Iancu" <<a moz-do-not-send="true"
href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>>
escreveu:
<div>
<div class="h5"><br type="attribution">
<blockquote class="gmail_quote" style="margin: 0pt 0pt
0pt 0.8ex; border-left: 1px solid rgb(204, 204,
204); padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000"> <tt>Take a
look at <a moz-do-not-send="true"
href="http://www.opensips.org/Resources/DocsTutRadius"
target="_blank">http://www.opensips.org/Resources/DocsTutRadius</a><br>
<br>
And be sure first that OpenSIPS (properly
configured) is sending the ACC request to the
RADIUS server.<br>
<br>
Regards,</tt><br>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<br>
On 12/18/2012 03:58 AM, Willian Mazzardo -
SYSSVOIP wrote:
<blockquote type="cite">
<p dir="ltr">Yes... I follow the tutorial in CDR
tool website.</p>
<p dir="ltr">There is any way to check if
everything is ok?<br>
</p>
<p dir="ltr">Thanks<br>
</p>
<div style="margin: 0pt 0pt 0pt 0.8ex;
border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000"> <tt>It
might be a silly question, but have you
configured the accounting via radius
backend ?<br>
<br>
Regards,</tt><br>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<br>
On 12/17/2012 11:49 PM, Willian Mazzardo -
SYSSVOIP wrote:
<blockquote type="cite">OK ... I have made
some tests and now I`m able to use
Dialplan module on Opensips-cp ... and are
working good.
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>How can I debug cdrtool to see what
is going on?</div>
<div><br>
</div>
<div>Thanks</div>
<div><br>
</div>
<div class="gmail_extra"><br clear="all">
<div>Willian Mazzardo<br>
Depto TI - SYSSVOIP<br>
<a moz-do-not-send="true"
href="http://www.syssvoip.com.br"
target="_blank">www.syssvoip.com.br</a><br>
<a moz-do-not-send="true"
href="tel:55%203537%202030"
value="+555535372030"
target="_blank">55 3537 2030</a></div>
<br>
<br>
<br>
<div class="gmail_quote">2012/12/17
Bogdan-Andrei Iancu <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:bogdan@opensips.org"
target="_blank">bogdan@opensips.org</a>></span><br>
<blockquote class="gmail_quote"
style="margin: 0pt 0pt 0pt 0.8ex;
border-left: 1px solid rgb(204, 204,
204); padding-left: 1ex;">
<div bgcolor="#ffffff"
text="#000000"> <tt>Hi Willian,<br>
<br>
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.<br>
<br>
Regards,<br>
</tt>
<pre cols="72">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a moz-do-not-send="true" href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a></pre>
<div>
<div> <br>
On 12/17/2012 08:19 PM,
Willian Mazzardo - SYSSVOIP
wrote:
<blockquote type="cite">Hi
Bogdan ... sorry for this
...
<div><br>
</div>
<div>I've initiated some
tests with Opensips ...
and almost everything is
working ...</div>
<div><br>
</div>
<div>Now, i`m trying do a
separate route for
internal accounts calls
and PSTN calls.</div>
<div><br>
</div>
<div>I`ve this script on
INVITE:</div>
<div>
<div><br>
</div>
<div> if
(is_method("INVITE")) {</div>
<div><br>
</div>
<div>
if(uri=~"^sip:[55910][0-9][0-9][0-9][0-9]@*")
{</div>
<div>
xlog("Willian: passou
por aqui PONTO A
PONTO");</div>
<div> route(3);</div>
<div><br>
</div>
<div> setflag(1); #
do accounting</div>
<div><br>
</div>
<div> }else{</div>
<div><br>
</div>
<div>
xlog("Willian: passou
por aqui SAIDA");</div>
<div><br>
</div>
<div>
rewritehostport("<a
moz-do-not-send="true"
href="http://177.126.178.106:5060" target="_blank">177.126.178.106:5060</a>");</div>
<div> route(5);</div>
<div><br>
</div>
<div> setflag(1); #
do accounting</div>
<div><br>
</div>
<div> }</div>
<div><br>
</div>
<div> setflag(1); #
do accounting</div>
<div> }</div>
</div>
<div><br>
</div>
<div>My internal accounts
start with 55910XXXX and
my PSTN calls are Country
Code + Region Code ...
like for Brazil = <a
moz-do-not-send="true"
href="tel:555588889999"
value="+555588889999"
target="_blank">555588889999</a></div>
<div><br>
</div>
<div>Is this INVITE section
right?</div>
<div><br>
</div>
<div>Thanks.</div>
<div><br>
</div>
<div><br>
</div>
<div class="gmail_extra"><br
clear="all">
<div>Willian Mazzardo<br>
Depto TI - SYSSVOIP<br>
<a
moz-do-not-send="true"
href="http://www.syssvoip.com.br" target="_blank">www.syssvoip.com.br</a><br>
<a
moz-do-not-send="true"
href="tel:55%203537%202030" value="+555535372030" target="_blank">55
3537 2030</a></div>
<br>
<br>
<br>
<div class="gmail_quote">2012/12/15
Bogdan-Andrei Iancu <span
dir="ltr"><<a
moz-do-not-send="true"
href="mailto:bogdan@opensips.org" target="_blank">bogdan@opensips.org</a>></span><br>
<blockquote
class="gmail_quote"
style="margin: 0pt 0pt
0pt 0.8ex;
border-left: 1px solid
rgb(204, 204, 204);
padding-left: 1ex;">
<div>
<div>Hi,</div>
<div><br>
</div>
<div>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</div>
<div><br>
</div>
<div>Regards,</div>
<div>Bogdan</div>
<div><br>
</div>
<div><br>
</div>
<div>
<div
style="font-size:
75%; color:
rgb(87, 87,
87);">Sent from
Samsung Mobile</div>
</div>
<div> <br>
Willian Mazzardo -
SYSSVOIP <<a
moz-do-not-send="true"
href="mailto:willian@syssvoip.com.br" target="_blank">willian@syssvoip.com.br</a>>
wrote:<br>
Hi all..
<div><br>
</div>
<div>I`m a very
new user coming
from Asterisk,
and I want to do
some test with
Kamailio billing
/ cdr my calls.</div>
<div><br>
</div>
<div>I have
installed
CDRTool and
Kamailio with a
working cfg who
route any call
to my SIP
Provider.</div>
<div><br>
</div>
<div>But, when I
do some call and
hang up later...
the system
doesn't create
any log into
radacct* tables.</div>
<div><br>
</div>
<div>I checked
every
configuration in
/etc/cdrtool/global.inc
and seems to be
OK.</div>
<div><br>
</div>
<div>I think maybe
is an kamailio
routing issue,
like no flag or
something.</div>
<div><br>
</div>
<div>Can anyone
help me with
this?</div>
<div><br>
</div>
<div>Thanks in
advice.</div>
<div><br>
</div>
<div><br
clear="all">
<div>Willian
Mazzardo<br>
Depto TI -
SYSSVOIP<br>
<a
moz-do-not-send="true"
href="http://www.syssvoip.com.br" target="_blank">www.syssvoip.com.br</a><br>
<a
moz-do-not-send="true"
href="tel:55%203537%202030" value="+555535372030" target="_blank">55
3537 2030</a></div>
<br>
</div>
</div>
</div>
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