On Tue, Mar 10, 2009 at 11:18 AM, Joshua Colp <jc...@digium.com> wrote: > ----- "Santiago Gimeno" <santiago.gim...@gmail.com> wrote: > This was filed as an issue and is being tracked at > http://bugs.digium.com/view.php?id=12437. Thus far > I have created a branch for Asterisk 1.4 that changes the behavior to accept > the incoming INVITE with > either audio and T38, or only T38 (if we only got T38). The outgoing call > initially goes out as audio until
In that case, Santiago should change the canreinvite=no to canreinvite=yes reload sip and see if that improves things. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
