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 <[email protected]> > ** > 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 > > > > 2012/12/15 Bogdan-Andrei Iancu <[email protected]> > >> 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 <[email protected]> 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 >> >> >
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