CHANUNAVAIL при специфичной команде звонка

Question

Доброго. Была в астериске Asterisk 1.8.13.1 такая запись:

exten => _8XXXXXXXXXX,n,Dial(SIP/gsm2/${EXTEN}:4006,,M(setAnswerDate))

Звонок шёл через GSM шлюз AddPac Model : GS1004_G2 на симку с hhotline = 4006 Был поднят другой сервер на замену этому, версии Asterisk 14.6.1, делаю так:

exten => _8XXXXXXXXXX,1,Answer
same => n,Dial(SIP/gsm2/${EXTEN}:4006,,U(subCallWriteAnswer,start,1,${EXTEN}))

Но получаю ошибку:

-- Executing [8MyNumber@internal:7] Dial("SIP/199-00000fd0", 

"SIP/gsm2/8MyNumber:4006,,U(subCallWriteAnswer,start,1,8MyNumber)") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/gsm2/8MyNumber:4006
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/199-00000fd0' status is 'CHANUNAVAIL'
    -- Executing [h@internal:1] NoOp("SIP/199-00000fd0", "INTERNAL HANGUP") in new stack

8MyNumber - скрыл свой номер просто. Если убираю 4006 то звонит нормально, но на любую свободную. Мне же надо конкретно на 4006. Интересно то, что я нигде не нашёл описания такого синтаксиса команды вообще, как это работало раньше не ясно. Симка в момент звонка свободная. Как позвонить через конкретную симку ?

Конфиг:

!
! APOS(tm) configuration saved from vty
!  2018/06/29 11:21:27 
!
version 8.51.004
!
hostname ITK-GSM2
clock timezone Irkutsk 9
!
username root password MYPASS administrator
username guest password guest user
!
!
script ntpdate default
 server ip pool.ntp.org
!
interface Loopback0
 ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
 ip address 192.168.221.12 255.255.255.192
 speed auto
 no qos-control
!
ip route 0.0.0.0 0.0.0.0 192.168.221.1 10
!
!
!
ftp server
ftp allow anonymous
http server
!
dns name-server XX.XXX.136.X
dns name-server XX.XXX.137.X
logging command
logging event 4-warning
logging on
! 
! 
! 
! 
! VoIP configuration. 
! 
! 
! Voice service voip configuration. 
! 
voice service voip 
 protocol sip
 dtmf-relay out-of-band
 fax protocol bypass 
 fax rate 9600 
 h323 call start fast 
 h323 call tunnel enable 
 no call-barring unconfigured-ip-address
 no voip-inbound-call-barring enable
! 
! 
! Voice port configuration. 
! 
! FXO 
voice-port 0/0 
 no caller-id enable 
! 
! 
! FXO 
voice-port 0/1 
 no caller-id enable 
! 
! 
! FXO 
voice-port 0/2 
 no caller-id enable 
! 
! 
! FXO 
voice-port 0/3 
 no caller-id enable 
! 
! 
! GSM 
voice-port 1/0 
 input gain 9 
 output gain 9 
 connection plar 4005 
 description line-1 
 ring detect-timeout 70 
 caller-id enable 
 caller-id type etsi 
 caller-id name disable
! 
! 
! GSM 
voice-port 1/1 
 input gain 9 
 output gain 9 
 connection plar 4006 
 description line-2 
 ring detect-timeout 70 
 caller-id enable 
 caller-id type etsi 
 caller-id name disable
! 
! 
! GSM 
voice-port 1/2 
 input gain 9 
 output gain 9 
 connection plar 4007 
 description line-3 
 ring detect-timeout 70 
 caller-id enable 
 caller-id type etsi 
 caller-id name disable
! 
! 
! GSM 
voice-port 1/3 
 input gain 9 
 output gain 9 
 connection plar 4008 
 description line-4 
 ring detect-timeout 70 
 caller-id enable 
 caller-id type etsi 
 caller-id name disable
! 
! 
! 
! 
! service port group configuration. 
! 
! 
! 
! Pots peer configuration. 
! 
dial-peer voice 1 pots
 destination-pattern .T 
 port 1/0 
 call-waiting
 preference 1
! 
dial-peer voice 2 pots
 destination-pattern .T 
 port 1/1 
 call-waiting
 preference 2
! 
dial-peer voice 3 pots
 destination-pattern .T 
 port 1/2 
 call-waiting
 preference 3
! 
dial-peer voice 4 pots
 destination-pattern .T 
 port 1/3 
 call-waiting
 preference 4
! 
! 
! 
! Voip peer configuration. 
! 
dial-peer voice 10 voip 
 destination-pattern 4T 
 session target ip ASTERISK_IP  5060 
 session protocol sip 
 voice-class codec 0 
 no vad
 dtmf-relay rtp-2833 
 fax protocol t38 redundancy 0 
 fax rate 9600 
! 
! 
! 
! 
! 
! 
gatekeeper
! 
! 
! Gateway configuration. 
! 
gateway 
 h323-id voip.192.168.221.12 
 no ignore-msg-from-other-gk 
! 
! 
! Codec classes configuration. 
! 
voice class codec 0 
 codec preference 1 g711alaw 
 codec preference 2 g711ulaw 
 codec preference 3 g729 
 codec preference 4 g7231r53 
 codec preference 5 g726r16 
 codec preference 6 g726r32 
! 
! 
! 
! SIP UA configuration. 
! 
sip-ua 
 user-register 
 sip-server ASTERISK_IP 
 register e164 
! 
! 
! Tones 
!
! 
! 
! 
! SMTP sendmail configuration 
! 
sms-delivery 
! 
! 
!
line console
!
line vty
 exec-timeout 0 0
!
gsm dev-restart-by-unreg 300
!
gsm 1/0
 sms-language utf8
!
gsm 1/1
 sms-language utf8
!
gsm 1/2
 sms-language utf8
!
gsm 1/3
 sms-language utf8
!