Depends on the configuration you make. For example, if you want to route the call giving the Mitel a new desrination or prefix, you can use Transfer dialplan app. Transfer before answering the call will be redirected with SIP 302.
If the call is to be anwered on *, then canreinvite set to yes or directrtp set to yes can help you. Saludos, Juan E. Rodríguez -----Original Message----- From: Kenneth Noisewater <[email protected]> Date: Thu, 1 Apr 2010 16:50:47 To: <[email protected]> Subject: [asterisk-users] SIP Connection Question -- _____________________________________________________________________ -- 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
