Addpac 1100. сигнал на FXS портах постоянно занято

Доброго времени суток.

Нужна помощь, готов отблагодарить.

Настраиваю AddPac AP1100-FXS8 в связке с Asterisk (FreePBX). Сделал настройки, подключаю аналоговый телефон к паку, на линии сигнал занято.. Соединение с астером проходит нормально:

AP1100# show dial-peer pots summary

 Tag     DestPattern        Prefix       Admin   Port      Regist  Inbound  Pref Hstop Tcalled Tcalling
-------------------------------------------------------------------------------------------------------
 1        405                             up     0/ 0       yes     -1        0  no     -1      -1

Вот настройки пака

!
version 8.12

! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
 fax protocol t38 redundancy 0
 fax rate 9600
 h323 call start fast
 timeout tidt 5
 timeout tras 8
 timing dtmf-duration 200
 local-ringback-tone early
 announcement language english
!
!
! Voice port configuration.
!
voice-port 0/0
! FXS
 input gain 1
 output gain 1
 no comfort-noise
 fax-early-detect
 no announcement
 caller-id enable

voice service voip
 minimize-voip-port multiply 1
!
! Pots peer configuration.
!
dial-peer voice 0 pots
 destination-pattern 405
 port 0/0
 no register e164
 user-password 568923voip
!
dial-peer voice 1 pots
 destination-pattern 405
 port 0/0
 user-password xxxxxx
!
!
!
! Voip peer configuration.
!
dial-peer voice 1001 voip
 destination-pattern .T
 session target 192.168.0.1
 session protocol sip
 codec g711alaw
 dtmf-relay rtp-2833
 no vad

gatekeeper
!
!
! Gateway configuration.
!
gateway
 h323-id voip.192.168.0.5
!
!
! SIP UA configuration.
!
sip-ua
 user-register
 sip-server 192.168.0.1
 timeout treg 3600
 register e164
!
!
! MGCP configuration.
!
mgcp
 epid-type
 codec  g711ulaw
!
!
! Tones
!
!
!

Привожу debug c пака

AP1100(config)# debug-port
AP1100(config)# 46      <Call   3>      : ******************  Call Created status(InitiatedByNet)  *******************
47      <SIP    3>      : Receive INVITE Request
48      <NetCon 3>      : Found inbound voip peer by dest-pattern id(1001)
49      <Call   3>      : From Net - calledParty(405) callingParty(309)
50      <Call   3>      : MatchedPerfect
51      <Call   3>      : MatchAllProcess After Sorted
                          <0>  id(1) dest(405) prefer(0) selected(1)
                          <1>  id(0) dest(405) prefer(0) selected(1)
52      <Call   3>      : Initiate callee with dial-peer(405) status(CalleeDeterminedAll) id(00000000-0000-0000-0000-000000000000)
53      <CEP    000000> : InitiateOutCall :  calledNum(), callingNum(309), callerPort(ffffffff) type(FXS)
54      <CEP    000000> : Outbound call to CEP callId(00000000-0000-0000-0000-000000000000) callNum(3)
55      <SIP    3>      : SetAlerting
56      <SIP    3>      : Receive CANCEL Request
57      <SIP    3>      : Cancel received
58      <Call   3>      : Terminated  from(fffffffe) this(Remote:CallClear) before(NULL) forced(0)
59      <CEP    000000> : StopSignal
60      <CEP    000000> : Disconnect (0)
61      <NetEP  3>      : Call TO <192.168.0.1> terminated reason(Remote:CallClear)
62      <SIP    3>      : Receive ACK Request
63      <CEP    000000> : Disconnected(16)

В какую сторону смотреть?, что нужно поменять?

Может кто готов помочь, готов отблагодарить.