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<p>Ok.</p>
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<p>Thank you!<br>
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<div class="PlainText">RODRIGO PIMENTA CARVALHO<br>
Inatel Competence Center<br>
Software<br>
Ph: +55 35 3471 9200 RAMAL 979 (Brasil)</div>
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<div id="divRplyFwdMsg" dir="ltr"><font style="font-size:11pt" face="Calibri, sans-serif" color="#000000"><b>De:</b> asterisk-users-bounces@lists.digium.com <asterisk-users-bounces@lists.digium.com> em nome de Yaron Nachum <nachum.yaron@gmail.com><br>
<b>Enviado:</b> segunda-feira, 20 de julho de 2015 02:53<br>
<b>Para:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Assunto:</b> Re: [asterisk-users] RES: Can I use PJSIP_HEADER to read the SIP 183 message header?</font>
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<div dir="ltr">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. <span style="font-size:13.1999998092651px; line-height:1.5">You will have
to dial again to the callee in the new extension.</span></div>
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<div dir="ltr"><span style="font-size:13.1999998092651px; line-height:1.5">This solution will increase the traffic on your asterisk and you have to be careful from loops.</span></div>
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<div dir="rtl">בתאריך יום ו׳, 10 ביולי 2015 ב-21:37 מאת Rodrigo Pimenta Carvalho <<a href="mailto:pimenta@inatel.br">pimenta@inatel.br</a>>:<br>
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Ok Mark Michelson.<br>
<br>
Thank you very much! You answer tells me that I was in the wrong path trying to access information from SIP 183 message.<br>
<br>
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.<br>
<br>
1- Is there any way to do that?<br>
<br>
2 - SIP MESSAGE, if sent by the calle, enters the dialplan?<br>
<br>
Any hint will be very helpful!<br>
<br>
Best regards.<br>
<br>
<br>
<br>
RODRIGO PIMENTA CARVALHO<br>
Inatel Competence Center<br>
Software<br>
Ph: +55 35 3471 9300 (Brasil)<br>
________________________________________<br>
De: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] em Nome de Mark Michelson
[<a href="mailto:mmichelson@digium.com" target="_blank">mmichelson@digium.com</a>]<br>
Enviado: sexta-feira, 10 de julho de 2015 15:14<br>
Para: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Assunto: Re: [asterisk-users] Can I use PJSIP_HEADER to read the SIP 183 message header?<br>
<br>
On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote:<br>
> Hi.<br>
><br>
> 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.<br>
><br>
> So, can I use PJSIP_HEADER to read the SIP 183 message header?<br>
><br>
> Any hint will be very helpful!<br>
><br>
> Best regards.<br>
><br>
><br>
> RODRIGO PIMENTA CARVALHO<br>
> Inatel Competence Center<br>
> Software<br>
> Ph: +55 35 3471 9200 RAMAL 979 (Brasil)<br>
Unfortunately, PJSIP_HEADER() cannot be used on responses because SIP<br>
responses do not enter the dialplan.<br>
<br>
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