[OpenSIPS-Users] Kamailio 3.1 + CDRTool + CallControl
Bogdan-Andrei Iancu
bogdan at opensips.org
Wed Dec 19 13:30:08 CET 2012
Hi Willian,
The packet type 15 is a Failure (in case of a failed/missed call) - and
this code is not officially supported by Freeradius. That is the patch
for :).
But first of all, do you see the ACC records in freeradius for
established calls ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 12/19/2012 02:46 PM, Willian Mazzardo - SYSSVOIP wrote:
> Hi again Bogdan ... thanks for your time and patience with this Noob
> ;) ehhehe
>
> I`m using now tarball of freeradius provided by AG (CDRTool) ...
>
> Now .. when I do some call ... this error appears:
>
> Wed Dec 19 08:44:10 2012 : Error: rlm_radutmp: NAS localhost port 5060
> unknown packet type 15)
>
> Googling this error, is about some patch to put into Freeradius ...
> but I dont know how to apply or where to find the patch.
>
>
> Is this error about patch ??
>
> Thank you
>
> Willian Mazzardo
> Depto TI - SYSSVOIP
> www.syssvoip.com.br <http://www.syssvoip.com.br>
> 55 3537 2030
>
>
> 2012/12/19 Bogdan-Andrei Iancu <bogdan at opensips.org
> <mailto:bogdan at opensips.org>>
>
> 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 <tel:55%203537%202030>
>>
>>
>>
>> 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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