Имеем следующее:
1) Asterisk
2) Cisco 2821 (c2800nm-adventerprisek9_ivs-mz.124-9.T.bin) 4 Voice FXO interfaces
3)  Eccom Basis
4) VoIP Сеть (в данном случай представителем будет Cisco IP Phone 7940)

Конфиги:

Asterisk

extensions.conf

exten => _8050X.,1,Dial(SIP/Cisco-1/70${EXTEN})
exten => _8067X.,1,Dial(SIP/Cisco-1/70${EXTEN})

sip.conf

[Cisco-1]                                                                                                                                                    
username=asterisk                                                                                                                                            
type=friend                                                                                                                                                  
secret=**********                                                                                                                                            
qualify=yes                                                                                                                                                  
nat=yes                                                                                                                                                      
insecure=very                                                                                                                                                
context=corp                                                                                                                                                
host=XXX.XXX.XXX.XXX                                                                                                                                          
disallow=all                                                                                                                                                
allow=alaw                                                                                                                                                  
allow=ulaw

Cisco

voice translation-rule 70
rule 1 /^708/ /8/
!
!
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 70
translate called 70
translate redirect-called 70

voice-port 0/1/1
input gain 10
output attenuation 10
connection plar opx 6555
description -==UMC==--
caller-id enable

dial-peer voice 500 pots
translation-profile incoming CALLER_ID_TRANSLATION_PROFILE
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
destination-pattern 7080.........
incoming called-number +380503550083
port 0/1/1
forward-digits all
prefix 70

В итоге имеем GSM-шлюз включённый в FXO Cisco и соединённый с ней астериск по SIP.

Трабл в следующем,
при звонке с IP-Phone на номер который попадает под вышеописаные правила слышим милый голос девушки оператора о неправильно набраном номере (догадка: цыска отдаёт не все цыфры шлюзу или первые просто не успевает передать)

Дальше шлюз сбрасывает девушку и слишим гудок который даёт шлюз набераем номер всё гуд.

Подскажите как это побороть?