[OpenSIPS-Users] Kamailio 3.1 + CDRTool + CallControl
Bogdan-Andrei Iancu
bogdan at opensips.org
Tue Dec 18 15:07:11 CET 2012
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 <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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