[Asterisk-Users] SIP Realtime Users

[Douglas Garstang]

Yusuf,
 
No I don't have the switch statement in extensions.conf. I'm not trying to do realtime extensions. I'm trying to do realtime SIP. They're different.
 
Doug.
 

	-----Original Message----- 
	From: yusuf [mailto:yusuf at ecntelecoms.com] 
	Sent: Sat 3/18/2006 6:49 AM 
	To: Asterisk Users Mailing List - Non-Commercial Discussion 
	Cc: 
	Subject: Re: [Asterisk-Users] SIP Realtime Users
	
	

	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
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