Ok Mark Michelson. Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message.
I need to find a way to let the callee pass information/data to the caller, even before accepting the call. That is, send data during the ringing time. And in my case, there will be more than one callee ringing at same time. As ASTERISK will not forward each SIP 183 message to the caller, I intend to get data from callees in dialplan by some another way before the call being accepted. 1- Is there any way to do that? 2 - SIP MESSAGE, if sent by the calle, enters the dialplan? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9300 (Brasil) ________________________________________ De: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] em Nome de Mark Michelson [mmichel...@digium.com] Enviado: sexta-feira, 10 de julho de 2015 15:14 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] Can I use PJSIP_HEADER to read the SIP 183 message header? On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote: > Hi. > > The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it > doesn't explain if such function works only over SIP INVITE messages or if it > can be use, for example, to read headers from others types of SIP messages > too. > > So, can I use PJSIP_HEADER to read the SIP 183 message header? > > Any hint will be very helpful! > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 (Brasil) Unfortunately, PJSIP_HEADER() cannot be used on responses because SIP responses do not enter the dialplan. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users