CHANUNAVAIL при специфичной команде звонка

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 !