I have an Asterisk Server behind a Fritz!Box.
All incoming lines from my provider are configured on the Fritzbox and set as SIP clients on the Asterisk like so:
[general] port = 5060 bindaddr = 0.0.0.0 language = de disallow = all allow = alaw allow = ulaw allow = gsm allow = g729 dtmfmode=auto fromdomain=10.10.10.254 register => 626Cisco:[SECRET]@10.10.10.254/626  type = peer username = 626Cisco secret = [SECRET] host = 10.10.10.254 fromdomain = fritz.box fromuser = 626Cisco nat = no canreinvite = yes insecure = port,invite  context = 10_SIP type = friend secret = [SECRET] host = dynamic nat = force_rport,comedia
On the Fritz!Box the MSN1 is bound to the extension “626”.
I spare out the rest, because the issue can be replicated by a single line object.
If I call from outside to the number of my MSN1 the call is routed to Asterisk and forwarded according to the dialplan:
[FritzBoxIncomingMSN01] ;Incoming Call on MSN1 (xxxxx) 60 sec ring, than voicebox exten => 626,n,dial(sip/10,60); exten => 626,n,VoiceMail(9); exten => 626,n,hangup();
The Fritz!Box has IP 10.10.10.254 and the Asterisk 10.10.10.242.
on the SIP Device registered with “10” this is displayed:
I can pickup the call, but I cannot call back if I miss, because the SIP Gateway is displayed from the Fritz!Box.
What do I need to change, that the caller is changed to