На сервере freepbx настроен SIP транк на провайдера телефонии. Содержимое транка
host=195.128.182.62
username=44xxxxxxx
fromuser=44xxxxxxx
secret=password
type=friend
insecure=port,invite
qualify=yes
dtmfmode=inband
diallow=all
context=from-trunk
canreinvite=no
allow=alaw&g729&ulaw
registerattempts=0
rtpkeepalive=10
fromdomain=86.111.xxx.xxx
nat=yes
Исходящие звонки работают нормально

при входящем звонке номер отбивается и выдаётся сообщение, что экстеншин не обслуживается. Все звонки завёрнуты на номер 101

[2012-11-08 10:37:49] NOTICE[10603]: chan_sip.c:22906 handle_response_peerpoke: Peer '101' is now Reachable. (7ms / 2000ms)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [s@from-trunk:1] NoOp("SIP/tr-InterTel-44xxxxxxx-00000000", "No DID or CID Match") in new stack
    -- Executing [s@from-trunk:2] Answer("SIP/tr-InterTel-44xxxxxxx-00000000", "") in new stack
    -- Executing [s@from-trunk:3] Wait("SIP/tr-InterTel-44xxxxxxx-00000000", "2") in new stack
    -- Executing [s@from-trunk:4] Playback("SIP/tr-InterTel-44xxxxxxx-00000000", "ss-noservice") in new stack
    -- <SIP/tr-InterTel-44xxxxxxx-00000000> Playing 'ss-noservice.gsm' (language 'en')
  == Spawn extension (from-trunk, s, 4) exited non-zero on 'SIP/tr-InterTel-44xxxxxxx-00000000'
    -- Executing [h@from-trunk:1] Macro("SIP/tr-InterTel-44xxxxxxx-00000000", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/tr-InterTel-44xxxxxxx-00000000", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("SIP/tr-InterTel-44xxxxxxx-00000000", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] Hangup("SIP/tr-InterTel-44xxxxxxx-00000000", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/tr-InterTel-44xxxxxxx-00000000' in macro 'hangupcall'
  == Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/tr-InterTel-44xxxxxxx-00000000'
aster*CLI>

Вот состояние пиров:

aster*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport ACL Port     Status      Description        
100/100                   192.168.27.100                           D              A  5060     OK (6 ms)                      
101/101                   192.168.27.7                             D              A  5060     OK (6 ms)                      
....
                      
tr-InterTel-44xxxxxxx/44x 195.128.182.62                               N             5060     OK (11 ms)
              
В inbound routes присутствует SIP_44xxxxxxx у которого обозначено destination 101.
Увы не работает.
Можете подсказать в чём засада?

root@aster:~/asterisk-11.0.1# asterisk -V
Asterisk 11.0.1
root@aster:~/asterisk-11.0.1#

FreePBX версия 2.10.1.1