Hi, thanks , i need to put this in the sip context...????
regards Upendra. On Fri, Jul 13, 2012 at 3:15 PM, Zohair Raza <engineerzuhairr...@gmail.com>wrote: > try with SipAddHeader(uri=answer-after=0) > > check syntax for Addheader > > Regards, > Zohair Raza > > > > > On Fri, Jul 13, 2012 at 1:42 PM, upendra <uppi...@gmail.com> wrote: > > Hi, > > > > > > I am trying to write dial plan for sip to auto answer (auto attend) the > > incoming call to the sip phone. > > > > - If i call from sip1 to sip2 then sip2 should automatically answer the > call > > and play some sound file. > > I am trying to do this but as new to the asterisk dial plan > configuration , > > so not able Todo this. > > help me if anyone already done this setup. > > > > > > > > Regards > > Upendra. > > > > -- > > _____________________________________________________________________ > > -- 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 >
-- _____________________________________________________________________ -- 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