Drop calls cause of missing extensions

Hi all!

I’ve installed asterisk( newest Freepbx on ubuntu 12.04. Now I made all configs as my provider told me:

1)Made new SIP trunk with this settings:

Trunk Name:Name
Outbound CallerID:2105050
PEER Details:
secret=my password

Register String:

I didnt touch any other settings in there.

2)Made an extension as a generic SIP device
Display Name:Nik
UserID: 100
Password: made simple pass
In Device Options: context: public
nat: no(rfc3581)
port: 5060
allow: alaw

No more changes since default

3)Made an incoming route
Description: in
Set Destination: Extensions 100

I entered no DID number or CallerID number.

4)Configured softphone:Downloaded X-Lite and made account changess:
Account name: Nik
Password: entered pass from extension
User ID = 100
Domain = stands there)
Display name:Nik
Auth. name = 100

Now I connected and everything seems to be OK.

I go to CLI and write sip show registry:

PBX*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
voice.melt.ru:5060 N 2105050 105 Registered Thu, 24 Oct 2013 08:32:19
1 SIP registrations.

then show peers

PBX*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
100/100 D A 29102 OK (47 ms)
Melt1/2105050 N 5060 OK (5 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

Then when i try to ring on my softphone
[2013-10-24 08:12:53] NOTICE[3386]: chan_sip.c:22622 handle_request_invite: Call from ‘2105050’ ( to extension ‘2105050’ rejected because extension not found in context ‘incoming’

What can I do to fix this?

This appears to be a FreePBX question. Support is provided at freepbx.org/forums
The problem is in extensions.conf, but you will need to manipulate this using the GUI.

Note that your trunk configuration contains the common set of questionable and deprecated values recommened by ISPs, but this wil not be causing your problem.

If you do need to bring this back to an Asterisk forum, please use Asterisk Support.