[OpenSIPS-Users] Kamailio 3.1 + CDRTool + CallControl

Willian Mazzardo - SYSSVOIP willian at syssvoip.com.br
Tue Dec 18 19:44:24 CET 2012


Hi Bogdan, It wasnt set aaa_flag ... and now is it.

Im trying install freeradius 1.1.3 from tarball ... and when I do make
command, this error appears:

.libs/modules.o: In function `setup_modules':
/usr/src/freeradius-1.1.3/src/main/modules.c:704: undefined reference to
`lt__PROGRAM__LTX_preloaded_symbols'
collect2: ld returned 1 exit status
make[4]: *** [radiusd] Error 1
make[4]: Leaving directory `/usr/src/freeradius-1.1.3/src/main'
make[3]: *** [common] Error 2
make[3]: Leaving directory `/usr/src/freeradius-1.1.3/src'
make[2]: *** [all] Error 2
make[2]: Leaving directory `/usr/src/freeradius-1.1.3/src'
make[1]: *** [common] Error 2
make[1]: Leaving directory `/usr/src/freeradius-1.1.3'
make: *** [all] Error 2


If i use debian freeradius package... this errors appears:

Tue Dec 18 16:35:13 2012 : Error: /etc/freeradius/sql.conf[21]:
Instantiation failed for module "sql"
Tue Dec 18 16:35:13 2012 : Error: /etc/freeradius/radiusd.conf[765]: Failed
to load module "sql".
Tue Dec 18 16:35:13 2012 : Error: /etc/freeradius/radiusd.conf[763]: Errors
parsing accounting section.
Tue Dec 18 16:35:13 2012 : Error: Failed to load virtual server <default>


Any help?


Willian Mazzardo
Depto TI - SYSSVOIP
www.syssvoip.com.br
55 3537 2030



2012/12/18 Bogdan-Andrei Iancu <bogdan at opensips.org>

> **
> 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 Developerhttp://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
> 55 3537 2030
>
>
>
> 2012/12/18 Willian Mazzardo - SYSSVOIP <willian at syssvoip.com.br>
>
>> Ok. I will do that.
>>
>> Thanks
>> Em 18/12/2012 05:06, "Bogdan-Andrei Iancu" <bogdan at opensips.org>
>> 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 Developerhttp://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 Developerhttp://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
>>> 55 3537 2030
>>>
>>>
>>>
>>> 2012/12/17 Bogdan-Andrei Iancu <bogdan at opensips.org>
>>>
>>>>  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 Developerhttp://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");
>>>>         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
>>>>
>>>>  Is this INVITE section right?
>>>>
>>>>  Thanks.
>>>>
>>>>
>>>>
>>>> Willian Mazzardo
>>>> Depto TI - SYSSVOIP
>>>> www.syssvoip.com.br
>>>> 55 3537 2030 <55%203537%202030>
>>>>
>>>>
>>>>
>>>> 2012/12/15 Bogdan-Andrei Iancu <bogdan at opensips.org>
>>>>
>>>>>  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 <willian at syssvoip.com.br> 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
>>>>> 55 3537 2030 <55%203537%202030>
>>>>>
>>>>>
>>>>
>>>
>
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