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[pimenta at inatel.br]

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-bounces@lists.digium.com [asterisk-users-bounces@lists.digium.com] em Nome de Mark Michelson [mmichelson@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:

[nachum.yaron at gmail.com]

One way to do it is to use Transfer. This will cause the callee to send 302 redirect to the caller. The caller then will jump to the extension specified in the contact. You will have to dial again to the callee in the new extension.


This solution will increase the traffic on your asterisk and you have to be careful from loops.






‫בתאריך יום ו׳, 10 ביולי 2015 ב-21:37 מאת ‪Rodrigo Pimenta Carvalho‬‏ <‪pimenta@inatel.br (pimenta@inatel.br)‬‏>:‬

[pimenta at inatel.br]

Ok.


Thank you!


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979 (Brasil)


De: asterisk-users-bounces@lists.digium.com <asterisk-users-bounces@lists.digium.com> em nome de Yaron Nachum <nachum.yaron@gmail.com>
Enviado: segunda-feira, 20 de julho de 2015 02:53
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] RES: Can I use PJSIP_HEADER to read the SIP 183 message header?

One way to do it is to use Transfer. This will cause the callee to send 302 redirect to the caller. The caller then will jump to the extension specified in the contact. You will have to dial again to the callee in the new extension.


This solution will increase the traffic on your asterisk and you have to be careful from loops.






‫בתאריך יום ו׳, 10 ביולי 2015 ב-21:37 מאת ‪Rodrigo Pimenta Carvalho‬‏ <‪pimenta@inatel.br (pimenta@inatel.br)‬‏>:‬