Hi Gustavo, I think one solution for this case is send and receive the audio during the early media (183). Asterisk when receive a ANM from pstn side not forward the 200 Ok to SIP side and establish the audio during the early media (183). Does anyone know if it is possible ? Regards, Bruno Rodrigues -------------------------------------------------- From: "Gustavo Marsico" <gustavomarsico@xxxxxxxxx> Sent: Friday, February 05, 2010 11:40 PM To: <asterisk-ss7 at lists.digium.com> Cc: <jvalencia at chile.com> Subject: Re: Charge indicator > Unfortunately Asterisk doesn't have any way to map the charge indicator in > SIP. Actually, there are a couple of drafts, but nothing serious at this > time. > If the other side supports it, you can send a P- or X- header to let the > other side knows if the call is chargeable or not. > > IMHO, in SIP terms, this is one of two biggest challenges for this > protocol. The other is the lack of support of SUSpend RESume capabilities > in the local loop side. > > Regards, > > Gustavo > > > On 5 Feb 2010, at 22:56, Bruno Rodrigues de Mello wrote: > >> Hi Jorge, >> >> My problem is not when I receive a call but when I send a call to telco. >> I use my asterisk box like a gateway and receive sip calls to route this >> calls to PSTN. >> In some cases the Telco send a ACM with charge indicator = 1 (no charge) >> and after this >> the telco send a ANM. >> When asterisk receive the ANM, it send a 200 Ok to SIP side and my >> softswitch start bill the call. >> >> Anyone has a idea ? >> >> Regards, >> Bruno Rodrigues >> >> >> >> From: Jorge Valencia >> Sent: Friday, February 05, 2010 6:20 PM >> To: asterisk-ss7 at lists.digium.com >> Subject: Re: [asterisk-ss7] Charge indicator >> >> >> Hi Bruno, well last year i had the same problem, it was posted here. My >> solution was modify the source, inside isup.c you need modify the code, >> find this function static FUNC_SEND(backward_call_ind_transmit) and add >> some lines. I think Matt was going to setup an option..anyway here is the >> code >> >> >> static FUNC_SEND(backward_call_ind_transmit) >> { >> unsigned char alwayscharge= 2; >> parm[0] = 0x40 | alwayscharge; >> parm[1] = 0x14; >> return 2; >> } >> >> Regards >> >> Jorge Valencia G. >> Operaciones >> Will Telefon?a SA >> Santo Domingo 1894 - Santiago - Chile >> +56 2 5720000 >> >> >> >> -------------------------------------------------------------------------------- >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-ss7 mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-ss7-- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-ss7 mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 >