[OpenSIPS-Users] Kamailio 3.1 + CDRTool + CallControl
Bogdan-Andrei Iancu
bogdan at opensips.org
Wed Dec 19 08:19:00 CET 2012
Hi Willian,
For the freeradius part, you should look into they documentation to see
why it fails to install. When using debs, it seems a config issue to me.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12/18/2012 10:44 PM, Willian Mazzardo - SYSSVOIP wrote:
> 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 <http://www.syssvoip.com.br>
> 55 3537 2030
>
>
>
> 2012/12/18 Bogdan-Andrei Iancu <bogdan at opensips.org
> <mailto: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 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 <tel:55%203537%202030>
>>
>>
>>
>> 2012/12/18 Willian Mazzardo - SYSSVOIP <willian at syssvoip.com.br
>> <mailto:willian at syssvoip.com.br>>
>>
>> Ok. I will do that.
>>
>> Thanks
>>
>> Em 18/12/2012 05:06, "Bogdan-Andrei Iancu"
>> <bogdan at opensips.org <mailto: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 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 <bogdan at opensips.org
>>>> <mailto: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 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
>>>>> <bogdan at opensips.org <mailto: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
>>>>> <mailto: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 <http://www.syssvoip.com.br>
>>>>> 55 3537 2030 <tel:55%203537%202030>
>>>>>
>>>>>
>>>>
>>
>
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