[Asterisk-Users] SIP Realtime Users

[Douglas Garstang]

Trying to get SIP realtime working here...

I'm connected to the database...

*CLI> realtime mysql status
Connected to vox180internal at db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds.

I can get information for the extension in question...

*CLI> realtime load sipusers name 2944093
                   Column Name  Column Value                  
          --------------------  --------------------          
                            id  1                             
                          name  2944093                       
                   accountcode  2944093                       
                     callgroup  1                             
                   canreinvite  no                            
                       context  From_OneEighty                
                      dtmfmode  auto                          
                           nat  rfc35                         
                   pickupgroup  1                             
                       qualify  no                            
                          type  friend                        
                      username  2944093                       
                      disallow  all                           
                         allow  g729                          
                         allow  ilbc                          
                         allow  gsm                           
                         allow  ulaw                          
                         allow  alaw                          
                    regseconds  0                             
                cancallforward  yes                           
              subscribecontext  sub_oneeighty                 

First of all, why doesn't Asterisk show _ALL_ the fields in the table? There's way more than this.

Second, when my phone comes up, asterisk displays this on the console:

*CLI> Mar 17 16:31:03 NOTICE[13354]: chan_sip.c:10854 handle_request_register: Registration from '<sip:2944093 at ipt.oneeighty.com>' failed for '216.xxx.142.205' - Username/auth name mismatch

I'm trying to do this in insecure mode, so Asterisk shouldn't even be asking the phone for a password. What's the deal? When I run an ngrep on the database, I can see that Asterisk isn't even TRYING to query the extension. Huh??? My sip.conf just has a [global] section, no users are provisioned in it. 

Doug.