On Sat, Sep 11, 2010 at 9:40 PM, Zeeshan Zakaria <[email protected]> wrote: > Poster is having problem when he disallows anonymous sip peers. Do you know > at all how FreePBX deals with anonymous sip peers? Obviously you haven't yet > seen the dialplan for FreePBX. > It's very simple to find the actually issue, if the OP does the following:
http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt The attached the debug log to thread. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: [email protected] | IRC: pabelanger (Freenode) blog.polybeacon.com -- _____________________________________________________________________ -- 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
