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.**** > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -- Thank you... Amar Akshat *"I am a C programmer. I do not program user interfaces unless they are console based. I read from stdin and a I write to stdout."* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20120322/a16babe8/attachment.html>
