Thanks for the prompt answer.
What do you mean is that for example
A calls B
I want B redirect the call to C.
So I have to call function pjsua_call_make_call from B and set the
destination to C, correct?
This way how A will know that the destination was changed to send RTP to it?
Also, is it possible to ring both lines simultaneously and answer from any
one of the two lines?
Thanks, Leonid.
From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org]
On Behalf Of Amar Akshat
Sent: Wednesday, March 21, 2012 6:13 PM
To: Leonid at myezconn.com; pjsip list
Subject: Re: Redirecting incoming call
I guess if you already have the redirection SIP URI defined, you could
simply
pjsua_call_make_call
to that destination_uri. So the intermediate node would generally receive
the INVITE, and call pjsua_call_make_call.
I can write sample code if you want.
Amar
On Thu, Mar 22, 2012 at 1:08 AM, Leonid Segal <voknetlab at gmail.com> wrote:
Hi All.
Is there any way to redirect an incoming call to another sip address?
Thanks.
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Amar Akshat
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