"Unable to find key"

I just unlocked a Linksys RT31P2 and set it up so that it should work with my Asterisk server. It’s not working however. The log (/var/log/asterisk/full) shows the following:

May 14 03:57:28 DEBUG[7956] db.c: Unable to find key ‘405’ in family 'SIP/Registry’
May 14 03:57:29 DEBUG[7956] chan_sip.c: Auto destroying call '640701e4-1795a463@192.168.10.151’
May 14 03:57:29 DEBUG[7956] db.c: Unable to find key ‘405’ in family 'SIP/Registry’
May 14 03:57:30 DEBUG[7956] chan_sip.c: Auto destroying call '753492d-fcc5e13@192.168.10.151’
May 14 03:57:30 DEBUG[7956] db.c: Unable to find key ‘405’ in family 'SIP/Registry’
May 14 03:57:31 DEBUG[7956] chan_sip.c: Auto destroying call '2c5f3bae-4558a5eb@192.168.10.151’
May 14 03:57:31 DEBUG[7956] db.c: Unable to find key ‘405’ in family 'SIP/Registry’
May 14 03:57:32 DEBUG[7956] chan_sip.c: Auto destroying call '1e0f70e7-5712ca6b@192.168.10.151’
May 14 03:57:32 DEBUG[7956] db.c: Unable to find key ‘405’ in family 'SIP/Registry’
May 14 03:57:33 DEBUG[7956] chan_sip.c: Auto destroying call ‘a77e3858-2e58c213@192.168.10.151’

What do I need to do to fix this?

Assuming the log comes from your Asterisk (i.e., not from Linksys), your dial plan involves AstDB. If this is not a vanilla Asterisk configuration, you need to post relevant information, including version, dial plan, etc.

The log I posted is from my Asterisk server. The setup is plain vanilla, version 1.2.17. I haven’t changed any dial plans, I’m able to connect fine with the same credentials using x-lite, but with the RT31P2 I get this error.

Then it makes sense to post things like sip.conf, and indicate which extension you tried to dial from Linksys.

I’m not dialing anything. As the log shows, the device is not registering properly. I don’t get a dial tone. In any case, heres my sip.conf:

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn’t, try adding “nat=1” to each peer definition to
; solve translation problems.

[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to ‘from-trunk’, rather than ‘from-sip-external’.
; You’ll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-trunk ; Send unknown SIP callers to this context
callerid = Unknown
tos=0x68

; #, in this configuration file, is NOT A COMMENT. This is exactly
; how it should be.
#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf
;#externhost=proxy.aiias.edu

You should have some of those included files? How did you register the soft phone? Also, what parameters are used in the Linksys to register?

The softphone registers using the same ID and password as I’ve set the RT31P2 up with. The only issue is that the RT31P2 is not registering. I haven’t changed any parameters on the Linksys, just set the server’s IP and the ID/password. What other files should I be posting? Asterisk has been working fine for months, the only thing that I’ve found is the error I posted previously, which I’d never gotten before.

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf