On Mon, Jun 2, 2014 at 4:14 PM, James Cloos <cl...@jhcloos.com> wrote: >>>>>> "MJ" == Matthew Jordan <mjor...@digium.com> writes: > > MJ> That is incorrect. The sip_sendhtml callback will update the url > MJ> stringfield on the SIP pvt. It then transmits a re-INVITE via > MJ> transmit_reinvite_with_sdp. > > There was no re-INVITE, just the initial INVITE. And it did not have an > Access-URL header. > > If Dial()'s url is only sent after the calle answers, it is of no value. > The callee needs the information to decide whether to answer.
That is what it does: * \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer So yes, the URL option probably isn't of much use to you. Whether or not it sent the re-INVITE: we'd have to investigate a lot further, which sounds like it isn't worth it. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev