I’ve installed asterisk(188.8.131.52)+ newest Freepbx on ubuntu 12.04. Now I made all configs as my provider told me:
1)Made new SIP trunk with this settings:
I didnt touch any other settings in there.
2)Made an extension as a generic SIP device
Password: made simple pass
In Device Options: context: public
No more changes since default
3)Made an incoming route
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 = 192.168.1.75(FreePBX stands there)
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
100/100 192.168.1.80 D A 29102 OK (47 ms)
Melt1/2105050 184.108.40.206 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: chan_sip.c:22622 handle_request_invite: Call from ‘2105050’ (220.127.116.11:5060) to extension ‘2105050’ rejected because extension not found in context ‘incoming’
What can I do to fix this?