Incomings are not routed without anydid/anicid route

Hi everyone
I have a tribox installed. I have a patton4960 with e1 interface. Outgoing calls works for now (with only problem the did not showing but maybe need patton work). Sip trunk for patton works unauthenticated with the following config
outbound caller id: PAtton <+30210211…>
peer details:
host=192.168.0.7
disallow=all
allow=alaw
qualify=yes
type=friend
context=from-trunk
canreinvite=yes

Everything else blank

Patton config
#----------------------------------------------------------------#

SN4960/1E30V/UI

R4.1 2007-05-24 H323 RBS SIP

2001-01-19T21:25:00

Generated configuration file

#----------------------------------------------------------------#

cli version 3.20
webserver port 80 language en

system

ic voice 0

system
clock-source 1 e1t1 0 0

profile ppp default

profile call-progress-tone defaultDialtone
play 1 1000 425 0

profile call-progress-tone defaultBusytone
play 1 480 425 -7
pause 2 480

profile call-progress-tone defaultReleasetone
play 1 240 425 -7
pause 2 240

profile call-progress-tone defaultCongestiontone
play 1 240 425 -7
pause 2 240

profile tone-set default

profile voip default
codec 1 g711alaw64k rx-length 20 tx-length 20
codec 2 g711ulaw64k rx-length 20 tx-length 20

profile pstn default

profile sip default

profile aaa default
method 1 local
method 2 none

context ip router

interface IF_IP_WAN
ipaddress dhcp
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

interface IF_IP_LAN
ipaddress 192.168.0.7 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu

context cs switch

routing-table called-e164 RT_ISDN_TO_SIP_0
route T2 dest-interface IF_SIP_0 MP_REM_CLIR

routing-table called-e164 RT_SIP_TO_ISDN_0
route default dest-interface IF_ISDN_0

mapping-table calling-pi to calling-e164 MP_REM_CLIR
map restricted to “”

interface isdn IF_ISDN_0
route call dest-table RT_ISDN_TO_SIP_0

interface sip IF_SIP_0
bind gateway GW_SIP_0
service default
route call dest-table RT_SIP_TO_ISDN_0
remote-party-id called-party
address-translation outgoing-call request-uri user-part fix 10000 host-part to-header target-param none
address-translation incoming-call called-e164 request-uri

context cs switch
no shutdown

gateway sip GW_SIP_0
bind interface IF_IP_LAN router

service default
domain 192.168.0.15
realm 192.168.0.15
defaultserver auto loose-router
registration auto 192.168.0.15
user 10000 authenticate password LbFCDNv4/Fk= encrypted

gateway sip GW_SIP_0
no shutdown

port ethernet 0 0
medium auto
encapsulation ip
bind interface IF_IP_WAN router
no shutdown

port ethernet 0 1
medium auto
encapsulation ip
bind interface IF_IP_LAN router
no shutdown

port e1t1 0 0
port-type e1
clock auto
framing crc4
encapsulation q921

q921
uni-side auto
encapsulation q931

q931
  protocol dss1
  uni-side user
  bchan-number-order ascending
  encapsulation cc-isdn
  bind interface IF_ISDN_0 switch

port e1t1 0 0
no shutdown

When anydid/anycid route exists all calls are routed to that extension. There are other routes but not working. When anydid/anycid route is deleted incoming calls fail (telco voice says number doesn’t exists) even if other routes exist
This is logged
[Mar 30 17:25:02] NOTICE[2486] chan_sip.c: Call from ‘10000’ to extension ‘10000’ rejected because extension not found.

Any ideas?