Thank you Klaus and Martin for your answers! It's very helpful! -- -- -- Marc LEURENT [email protected]
Le vendredi, 23 octobre 2009 20.51:54, Martin a écrit : > 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 > _______________________________________________ -- 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
