> When I dial into Asterisk ( I have a SIP trunk - which I need to make > sure is not faulty), I only get one-way voice communication. > The calling party, from the SIP trunk hears nothing - the extension > rings on the Asterisk server (you can see it in the CLI and hear it at > the computer), and the softphone rings
Try canredirect=no on both your sip peer and your sip provider. -- _____________________________________________________________________ -- 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
