[Problem] asterisk incoming/outgoing & fritzbox

[atux_null]

i do have a raspberry that acts as my PBX and it has asterisk. it is pure asterisk, no Freepbx.
i have a fritzbox 7270 as my router(192.168.178.1) that has the registration of the ITSP lines and the ISDN. I have a msn number 123456789 that i would like to send it to asterisk by creating an IP phone to fritzbox and registering it to asterisk.
Even though the number is registered i cannot make/receive calls:

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.178.1:5060;branch=z9hG4bK9FFECF83EA86E14B;received=192.168.178.1;rport=5060
From: <sip:[email protected]>;tag=06932390B6426E7B
To: "123456789" <sip:[email protected]:5060>;tag=as3c62f79e
Call-ID: [email protected]
CSeq: 16 INVITE
Server: Asterisk PBX 11.25.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[Sep  5 09:41:31] NOTICE[3328][C-00000007]: chan_sip.c:25904 handle_request_invite: Call from '620' (192.168.178.1:5060) to extension '620' rejected because extension not found in context 'Dial-In'.
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.178.1:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.1:5060;branch=z9hG4bK9FFECF83EA86E14B
From: <sip:[email protected]>;tag=06932390B6426E7B
To: "123456789" <sip:[email protected]:5060>;tag=as3c62f79e
Call-ID: [email protected]
CSeq: 16 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 7270 v3 74.05.53 (Feb 11 2014)
Content-Length: 0

the following are the settings from:
extensions.conf

[from-fritz]
exten => 620,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => 620,n,Answer()
exten => 620,n,Wait(1)
exten => 620,n,Dial(SIP/450,20)
exten => 620,n,VoiceMail(450@Office,su)
exten => 620,n,Busy(3)

and from
sip.conf

register => 620:[email protected]/620

[620]
context=from-fritz
type=peer
insecure=very
qualify=yes
dtmf=rfc2833
username=620
secret=123456789
host=192.168.178.1
disallow=all
allow=alaw
allow=g729
allow=ulaw

what do i have wrong? how can i alter the config to make outbound calls?

 

[rentier-s]

Da ich die Vorgehensweisen des neuen Forum-Betreibers nicht gut heiße, stehen meine Beiträge hier nicht weiter zur Verfügung.

 

[atux_null]

fixed. now it is coming to context from-fritz, which is correct.
Thanks

 

[atux_null]

Incoming works fine. The problem with the outgoing still exists.
450 is the extension number of my softphone, 8 is the seizure code to dial through that trunk.

  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [82108196487@AllCalls:1] Goto("SIP/450-00000003", "DefaultPlan,865478523546,1") in new stack
    -- Goto (DefaultPlan,865478523546,1)
    -- Executing [865478523546@DefaultPlan:1] Dial("SIP/450-00000003", "SIP/620/865478523546") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/620/865478523546
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [865478523546@DefaultPlan:2] Hangup("SIP/450-00000003", "1") in new stack
  == Spawn extension (DefaultPlan, 865478523546, 2) exited non-zero on 'SIP/450-00000003'
voyage*CLI>

extensions.conf

exten =>_8X.,1,Dial(SIP/620/${EXTEN})
exten =>_8X.,n,Hangup(${HANGUPCAUSE})

 

[rentier-s]

Da ich die Vorgehensweisen des neuen Forum-Betreibers nicht gut heiße, stehen meine Beiträge hier nicht weiter zur Verfügung.