СвязьПроект

Здравствуйте. Есть вопрос. На AddPac 1002 (APOS 8.51.002) при звонке на GSM порт звонок пробрасывается по IP trunk (H.323) на Avaya IP Office. В глосовом приветствии требуется DTMF донабор. При любых настройках DTMF донабор срабатывает не всегда.
В Debug voice call при срабатывании видны строки в виде:
При не работе:

В какую сторону копать?

Присоединяюсь к вопросу. Только вместо Avaya стоит Panasonic NCP500. AddPac подключен так же по H.323

Попробуйте обновить прошивку
http://www.4shared.com/file/tOp-dKJ4/gs ... 1_004.html

Эта прошивка уже стоит...
Я сейчас уже и не уверен что подглючивает именно шлюз, а не АТС...

Вот конфиг шлюза:

!

! APOS(tm) configuration saved from vty


!
!

hostname GSM1

clock timezone EKR 5

!

username root password router administrator

username guest password guest user

!

!

script ntpdate default

server ip time.windows.com

!

interface Loopback0
ip address 127.0.0.1 255.0.0.0

!

interface FastEthernet0/0

ip address 192.168.80.61 255.255.255.0

speed auto

no qos-control

!

interface FastEthernet0/1

no ip address

speed auto

no qos-control

!

interface FastEthernet0/1:1

ip address 192.168.10.1 255.255.255.0

!

ip route 0.0.0.0 0.0.0.0 192.168.80.150 10

!

!

!

ip dhcp pool default

network FastEthernet0/1

range FastEthernet0/0

subnet-mask 255.255.255.0

!

!
ftp server
http server
!

dns name-server 192.168.80.150

ip connect default inside FastEthernet0/1 outside FastEthernet0/0
!

logging command
logging event 4-warning
logging on
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
protocol h323
dtmf-relay out-of-band
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
busyout monitor gatekeeper
busyout monitor voip-interface
no call-barring unconfigured-ip-address
no voip-inbound-call-barring enable
!
!
! Voice port configuration.
!
! GSM
voice-port 0/0
connection plar 79194423891
caller-id enable
!
!
! GSM
voice-port 0/1
connection plar 79129886569
caller-id enable
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 3048 pots
port 0/0
!
dial-peer voice 3049 pots
port 0/1
!
dial-peer voice 4584 pots
destination-pattern 0T
port 0/0
no register e164
translate-outgoing called-number 300
!
dial-peer voice 4585 pots
destination-pattern 1T
port 0/1
no register e164
translate-outgoing called-number 300
!
!
!
! Voip peer configuration.
!
dial-peer voice 10000 voip
destination-pattern 79194423891
session target ip 192.168.80.60
answer-address T
voice-class codec 0
no vad
dtmf-relay h245-alphanumeric
!
dial-peer voice 10001 voip
destination-pattern 79129886569
session target ip 192.168.80.60
answer-address T
voice-class codec 0
no vad
dtmf-relay h245-alphanumeric
!
!
!
!
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.80.61
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
!
!
!
! Translation Rule configuration.
!
translation-rule 300
rule 0 T %02%99
!
translation-rule 10100
rule 0 T %01%99
!
!
!
! SIP UA configuration.
!
sip-ua
!
!
! Tones
!
!
!
!
line console

!
line vty

!
gsm 0/0
sms-language utf8

!
gsm 0/1
sms-language utf8

!

Вот что появляется в логе при неудачном звонке (после дозвона на DISA набрали вн. номер 121, звонок же ушел на 104)

75 <Chan 61> : Open - number(101) direction(transmit) session(voice)
codec(g711alaw-64k)
- Local : Data(23122) Cont(23123) Addr(192.168.80.61)
- Remote : Data(12054) Cont(12055) DataAddr(192.168.80
.63) ContAddr(192.168.80.63)
76 <Q931 61> : Received FACILITY
77 <H245 61> : Received TCS request.
78 <H245 61> : remote capabilities matching to local capabilities.
number of capabilities = 4
1 : g711alaw-64k
2 : g711ulaw-64k
3 : g729-8k
4 : T.38
79 <H245 61> : Send TCS ack.
80 <H245 61> : Received MSD request.
81 <H245 61> : Send MSD ack(master).
82 <H245 61> : Received TCS ack..
83 <H245 61> : Received MSD ack(slave).
84 <Call 61> : Digit(1) at Established
85 <H245 61> : Send UserInputTone(1) dur(0)
86 <Call 61> : Digit(2) at Established
87 <H245 61> : Send UserInputTone(2) dur(0)
88 <Call 61> : Digit(1) at Established
89 <H245 61> : Send UserInputTone(1) dur(0)
[152425.344] GSM-0/1: CALL DISCONNECT by DISCONNECT event
90 <CEP 000100> : Disconnected(16) at Busy
91 <Call 61> : Terminated from(100) this(Local:CallClear) before((nul
l)) forced(0) time(1314352820)
92 <CEP 000100> : DisconnectCall at Idle
93 <Chan 61> : Close - number(129) direction(receive)
94 <Chan 61> : Close - number(101) direction(transmit)
95 <Q931 61> : Send RELEASE COMPLETE
96 <NetEP 61> : Call TO <192.168.80.60> terminated reason(Local:CallCl
ear)
[152425.709] Interface GSM-0/1, changed state to DOWN
97 <Port 000100> : LinkDown
[152430.709] Interface GSM-0/1, changed state to UP
98 <Port 000100> : LinkUp