This might help to answer poster's question. It tells how the allow anonymous sip connections work in FreePBX, and shows the code.
http://www.geekzone.co.nz/sbiddle/7183 <http://www.geekzone.co.nz/sbiddle/7183>-- Zeeshan On Sun, Sep 12, 2010 at 12:11 AM, Paul Belanger < [email protected]> wrote: > 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 > -- Zeeshan A Zakaria
-- _____________________________________________________________________ -- 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
