Есть cisco 1750 с модулем 2FXO, голосовой платой + увеличенной флэш и опертивной памятью(со стандартной памятью голосовые прошивки не поставить). Пытаюсь настроить как голосовой шлюз эту железяку, но при попытке позвонить линия всегда занята. В лог астериска ничего не падает, логично предположить что звонок не проходит через циску. С чем может быть связанно?
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: cisco conf
Current configuration : 2796 bytes
!
! Last configuration change at 16:36:36 IRKT Tue Jun 17 2014
! NVRAM config last updated at 16:37:35 IRKT Tue Jun 17 2014
!
version 12.3
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname VGate
!
boot-start-marker
boot-end-marker
!
logging cns-events warnings
!
memory-size iomem 25
clock timezone IRKT 9
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
aaa new-model
!
!
aaa session-id common
ip subnet-zero
ip cef
!
!
!
ip name-server 91.219.136.4
ip name-server 91.219.137.4
no ftp-server write-enable
!
voice echo-canceller extended
voice rtp send-recv
!
voice service pots
supported-language ru
!
voice service voip
fax protocol pass-through g711alaw
sip
bind all source-interface FastEthernet0
min-se 60
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
!
!
!
voice class busyout 1
!
!
voice class dualtone-detect-params 1
cadence-variation 50
!
!
voice class custom-cptone RU
dualtone busy
frequency 570 640
cadence 500 500
!
!
!
!
!
voice translation-profile VoiceTraPro
!
!
!
!
gw-accounting aaa
!
!
!
!
interface FastEthernet0
ip address 192.168.2.128 255.255.255.0
shutdown
speed auto
!
interface Serial0
no ip address
shutdown
!
interface Serial1
no ip address
shutdown
!
ip default-gateway 192.168.2.1
ip classless
no ip http server
!
!
voice-port 1/0
no comfort-noise
cptone RU
timeouts call-disconnect 1
timeouts wait-release 1
timing hookflash-out 1300
timing digit 200
timing inter-digit 200
timing guard-out 500
connection plar 5001
supervisory disconnect dualtone mid-call
supervisory dualtone-detect-params 1
no battery-reversal
!
voice-port 1/1
no comfort-noise
cptone RU
timeouts call-disconnect 1
timeouts wait-release 1
timing hookflash-out 1300
timing digit 200
timing inter-digit 200
timing guard-out 500
connection plar 5002
supervisory disconnect dualtone mid-call
supervisory dualtone-detect-params 1
no battery-reversal
!
mgcp
!
!
dial-peer voice 1 pots
destination-pattern .T
no digit-strip
direct-inward-dial
port 1/0
forward-digits 0
!
dial-peer voice 2 pots
destination-pattern .T
no digit-strip
direct-inward-dial
port 1/1
forward-digits 0
!
dial-peer voice 10 voip
huntstop
destination-pattern 5T
progress_ind setup enable 3
voice-class sip url sip
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
codec g711alaw
fax protocol pass-through g711alaw
no vad
!
sip-ua
calling-info sip-to-pstn name set SIPTOPSTN
max-forwards 15
dtmf-relay rtp-nte
codec g711alaw
fax protocol pass-through g711alaw
no vad
!
sip-ua
calling-info sip-to-pstn name set SIPTOPSTN
max-forwards 15
sip-server ipv4:91.219.137.26:5060
!
!
line con 0
line aux 0
line vty 0 4
exec-timeout 60 0
logging synchronous
transport input telnet
!
ntp master 3
ntp server 192.168.128.6 source FastEthernet0
!
end
!
! Last configuration change at 16:36:36 IRKT Tue Jun 17 2014
! NVRAM config last updated at 16:37:35 IRKT Tue Jun 17 2014
!
version 12.3
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
!
hostname VGate
!
boot-start-marker
boot-end-marker
!
logging cns-events warnings
!
memory-size iomem 25
clock timezone IRKT 9
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
aaa new-model
!
!
aaa session-id common
ip subnet-zero
ip cef
!
!
!
ip name-server 91.219.136.4
ip name-server 91.219.137.4
no ftp-server write-enable
!
voice echo-canceller extended
voice rtp send-recv
!
voice service pots
supported-language ru
!
voice service voip
fax protocol pass-through g711alaw
sip
bind all source-interface FastEthernet0
min-se 60
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
!
!
!
!
voice class busyout 1
!
!
voice class dualtone-detect-params 1
cadence-variation 50
!
!
voice class custom-cptone RU
dualtone busy
frequency 570 640
cadence 500 500
!
!
!
!
!
voice translation-profile VoiceTraPro
!
!
!
!
gw-accounting aaa
!
!
!
!
interface FastEthernet0
ip address 192.168.2.128 255.255.255.0
shutdown
speed auto
!
interface Serial0
no ip address
shutdown
!
interface Serial1
no ip address
shutdown
!
ip default-gateway 192.168.2.1
ip classless
no ip http server
!
!
voice-port 1/0
no comfort-noise
cptone RU
timeouts call-disconnect 1
timeouts wait-release 1
timing hookflash-out 1300
timing digit 200
timing inter-digit 200
timing guard-out 500
connection plar 5001
supervisory disconnect dualtone mid-call
supervisory dualtone-detect-params 1
no battery-reversal
!
voice-port 1/1
no comfort-noise
cptone RU
timeouts call-disconnect 1
timeouts wait-release 1
timing hookflash-out 1300
timing digit 200
timing inter-digit 200
timing guard-out 500
connection plar 5002
supervisory disconnect dualtone mid-call
supervisory dualtone-detect-params 1
no battery-reversal
!
mgcp
!
!
dial-peer voice 1 pots
destination-pattern .T
no digit-strip
direct-inward-dial
port 1/0
forward-digits 0
!
dial-peer voice 2 pots
destination-pattern .T
no digit-strip
direct-inward-dial
port 1/1
forward-digits 0
!
dial-peer voice 10 voip
huntstop
destination-pattern 5T
progress_ind setup enable 3
voice-class sip url sip
session protocol sipv2
session target sip-server
session transport udp
dtmf-relay rtp-nte
codec g711alaw
fax protocol pass-through g711alaw
no vad
!
sip-ua
calling-info sip-to-pstn name set SIPTOPSTN
max-forwards 15
dtmf-relay rtp-nte
codec g711alaw
fax protocol pass-through g711alaw
no vad
!
sip-ua
calling-info sip-to-pstn name set SIPTOPSTN
max-forwards 15
sip-server ipv4:91.219.137.26:5060
!
!
line con 0
line aux 0
line vty 0 4
exec-timeout 60 0
logging synchronous
transport input telnet
!
ntp master 3
ntp server 192.168.128.6 source FastEthernet0
!
end
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: dial plan
[from-fxo]
exten => _5XXX,1,Answer
exten => _5XXX,n,GoSub(subInitVariables,start,1())
exten => _5XXX,n,GoSub(subCallWrite,start,1(5))
exten => _5XXX,n,GoSub(subStartCallMonitorIfNeed,start,1(0))
exten => _5XXX,n,Goto(itk-fromabonent,s,1)
[itk-fromabonent]
exten => s,1,Background(firstsound,)
exten => s,n,NoOp(CallerID(num) is ${CALLERID(num)} CallerID(name) is ${CALLERID(name)})
exten => s,n,GoTo(startqueue)
exten => s,n(startqueue),Macro(incall)
exten => i,1,Macro(incall)
exten => t,1,Macro(incall)
exten => h,1,NoOp(TEST)
exten => h,n,GoSub(subEndCallWrite,start,1())
exten => h,n,System(/var/www/protected/yiic "calls" "current" --id=${MYSQL_PARENT_CALL_ID} --hot=0)
;exten => h,n,AGI(/usr/local/bin/asterisk/billing_notif.php,${MYSQL_PARENT_CALL_ID},0)
exten => _5XXX,1,Answer
exten => _5XXX,n,GoSub(subInitVariables,start,1())
exten => _5XXX,n,GoSub(subCallWrite,start,1(5))
exten => _5XXX,n,GoSub(subStartCallMonitorIfNeed,start,1(0))
exten => _5XXX,n,Goto(itk-fromabonent,s,1)
[itk-fromabonent]
exten => s,1,Background(firstsound,)
exten => s,n,NoOp(CallerID(num) is ${CALLERID(num)} CallerID(name) is ${CALLERID(name)})
exten => s,n,GoTo(startqueue)
exten => s,n(startqueue),Macro(incall)
exten => i,1,Macro(incall)
exten => t,1,Macro(incall)
exten => h,1,NoOp(TEST)
exten => h,n,GoSub(subEndCallWrite,start,1())
exten => h,n,System(/var/www/protected/yiic "calls" "current" --id=${MYSQL_PARENT_CALL_ID} --hot=0)
;exten => h,n,AGI(/usr/local/bin/asterisk/billing_notif.php,${MYSQL_PARENT_CALL_ID},0)
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: sip conf
[cisco]
directmedia=no
type=friend
context=from-fxo
dtmfmode=info
host=192.168.2.128
permit=0.0.0.0/0.0.0.0
call-limit=4
canreinvite=no
disallow=all
allow=alaw
directmedia=no
type=friend
context=from-fxo
dtmfmode=info
host=192.168.2.128
permit=0.0.0.0/0.0.0.0
call-limit=4
canreinvite=no
disallow=all
allow=alaw
PS. Извиняюсь за тонны конфигов. Просто не знаю что может быть важно, а что не очень.
Вроде теперь вызовы проходят, но номера не определяются. Вместо номера выводит "FXO". Вроде должно все определяться, но что-то не получается. Сразу прошу простить за некоторые глупые моменты. Первый раз в жизни циску настраиваю, по мануалам все. Любым советам буду рад.
