[Asterisk-Users] SIP Realtime Users

[yusuf]

Douglas Garstang wrote:
> 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  



>                       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.
> 
> 
Hi,
do you have in sip.conf
[From_OneEighty]
switch => Realtime/sipusers at extensions