On Tue, Nov 9, 2010 at 6:35 PM, Brett Woollum <[email protected]> wrote: > Nobody has any idea why the Caller ID is being overwritten when using > Asterisk Realtime for the SIP users? > No, perhaps you can _show_ us the problem.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: [email protected] | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _____________________________________________________________________ -- 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
