You can call application Progress() from within dialplan and it will cause the Asterisk to send a SIP reply 183 on the call that came to Asterisk.
Martin On Fri, Oct 23, 2009 at 6:36 AM, Marc Leurent <[email protected]> wrote: > Hello everybody, > I have 2 users connected on the same Asterisk server that are connected with > 2 different Asterisk servers. > > For outgoing calls, one in receiving 183 Session Progress and the other not! > Do you have any idea why? > Thanks. > > I have tried to understand by myself and in their INVITE they have almost the > same Allow and Supported SIP Headers > > The one that works: > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO > Supported: replaces, timer > > The one that doen't work: > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > > -- > -- -- > Marc LEURENT > [email protected] > > _______________________________________________ > -- 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 > _______________________________________________ -- 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
