Rejected because extension not found

Guys I have installed asterisk 4.1 on my qnap device and am having issues calling out (both uk landline and mobile numbers)

My sipgate trunk is showing registered and my 3cx on my windows desktop showing registered to the PBX.

When I call an external number I get the an error on the softphone of “SERVER UNREACHABLE”

On the Asterisk logs I am seeing

[Dec 13 22:22:18] NOTICE[7464] chan_sip.c: Call from ‘’ to extension ‘99900972595301124’ rejected because extension not found.
[Dec 13 22:22:29] ERROR[6037] config.c: *********************************************************
[Dec 13 22:22:29] ERROR[6037] config.c: *********** YOU SHOULD REALLY READ THIS ERROR ***********
[Dec 13 22:22:29] ERROR[6037] config.c: Future versions of Asterisk will treat a #include of a file that does not exist as an error, and will fail to load that configuration file. Please ensure that the file ‘…/zaptel.conf’ exists, even if it is empty.
[Dec 13 22:22:29] ERROR[6037] config.c: *********** YOU SHOULD REALLY READ THIS ERROR ***********
[Dec 13 22:22:29] ERROR[6037] config.c: *********************************************************
[Dec 13 22:22:29] WARNING[6053] app_system.c: Unable to execute ‘ztscan > /etc/asterisk/ztscan.conf’
[Dec 13 22:39:11] NOTICE[7464] chan_sip.c: Call from ‘’ to extension ‘9900972595301124’ rejected because extension not found.
[Dec 13 22:55:46] NOTICE[7464] chan_sip.c: Call from ‘’ to extension ‘972595301124’ rejected because extension not found.
[Dec 13 22:56:02] WARNING[26586] channel.c: No channel type registered for ‘’
[Dec 13 22:56:02] WARNING[26586] app_dial.c: Unable to create channel of type ‘’ (cause 66 - Channel not implemented)

my sip.conf

context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[general]
static = yes
writeprotect = no
clearglobalvars = no

[globals]
CONSOLE = Console/dsp
IAXINFO = guest
TRUNK = Zap/G2
TRUNKMSD = 1
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =
PAGING_HEADER = Intercom
PAGING_TIMEOUT = 60
GLOBAL_OUTBOUNDCID =
GLOBAL_OUTBOUNDCIDNAME =
CID_6005 = 08451547038
2823768 = SIP/2823768
exten => 2823768,1,Playback(vm-goodbye)
exten => 2823768,n,Hangup()
08444879795 = SIP/08444879795
;TRUNK=IAX2/user:[email protected]

I am new to asterisk, please help me understand whats going on here its been 3 days and I am not getting any closer. Assistance please and thank you

Sorry from [general] is my extensions.conf

There is no asterisk 4.1, further your box does not look like it is using FreePBX (whose support forum you are in) further, zaptel was replaced by dahdi years ago further yet your config files are WAY out of line with a workable raw asterisk deployment.

I suggest you spend some time on the wiki here and then start over with something that you will find more understandable.

1.4 is the version in question.

The version is installed from the qnap app center, this is the version they recommend running on top of the nas firmware (I know its not the latest)

The files are pretty much untouched as I have only created a trunk, had it registered, created a dialplan and setup a single extension so I am not sure why they are not workable…

Thank you for your quick reply, chow

Then as I say you are in FreePBX land which is a frontend for asterisk, you will have to get support from your vendor, not here, sorry.