Помогите пожалуйста связать IPO500v2 и GS1500 по SIP-транку.
Схема следующая: в адпаке стоит симка, на него приходит звонок, который должен пройти на IPO. С исходящими ситуация обратная.
Исходящие вызовы работают а вот входящие нет.
Конфиг адпака:
Код: Выделить всё
Current configuration:
!
version 8.51.005
!
hostname addpac-avk-01
!
username root password router administrator
username guest password guest user
!
!
fan-control on
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 192.168.10.11 255.255.255.0
speed auto
no qos-control
!
interface FastEthernet0/1
ip address 192.168.1.11 255.255.255.0
shutdown
speed auto
no qos-control
!
!
!
http server
!
!
logging command
logging event 4-warning
logging on
!
!
!
!
! VoIP configuration.
!
!
! Controller configuration.
!
!
!
! Voice service voip configuration.
!
voice service voip
protocol sip
dtmf-relay out-of-band
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
busyout monitor sip-server
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 89501542114
!
!
! GSM
voice-port 0/1
!
!
! GSM
voice-port 0/2
!
!
! GSM
voice-port 0/3
!
!
!
!
! service port group configuration.
!
!
!
! Pots peer configuration.
!
dial-peer voice 1 pots
destination-pattern T
port 0/0
user-name addpac
user-password 117133
!
!
!
! Voip peer configuration.
!
dial-peer voice 2 voip
destination-pattern T
session target sip-server
session protocol sip
voice-class codec 1
no vad
dtmf-relay rtp-2833
!
!
!
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
h323-id voip.192.168.10.13
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g711ulaw
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729
codec preference 4 g7231r63
codec preference 5 g7231r53
!
!
!
! SIP UA configuration.
!
sip-ua
fault-tolerance 3 500
user-register
sip-server 192.168.10.13
retry-counter 3
remote-party-id
register e164
!
!
! Tones
!
!
!
!
line console
!
line vty
!
gsm dev-restart-by-unreg 300
!
gsm 0/0
sms-language utf8
cdma-sms-language unspecify encoding ascii
!
gsm 0/1
sms-language utf8
cdma-sms-language unspecify encoding ascii
!
gsm 0/2
sms-language utf8
cdma-sms-language unspecify encoding ascii
!
gsm 0/3
sms-language utf8
cdma-sms-language unspecify encoding ascii
!
gsm 1/0
sms-language utf8
cdma-sms-language unspecify encoding ascii
!
gsm 1/1
sms-language utf8
cdma-sms-language unspecify encoding ascii
!
gsm 1/2
sms-language utf8
cdma-sms-language unspecify encoding ascii
!
gsm 1/3
sms-language utf8
cdma-sms-language unspecify encoding ascii
!
end
Настройка линии на IPO: Скрин кладки Учётные данные SIP почему-то не даёт вставить.
Та заведена учётная запись с параметрами как в конфиге адпака.
Вывод show sip с адпака:
Код: Выделить всё
Proxyserver Registration Information
proxyserver registration option = e164
Proxyserver list :
---------------------------------------------------------------------------
Server address Port Priority Domain Status(LastFailReason)
---------------------------------------------------------------------------
192.168.10.13 5060 128 any Failed(Rx:OtherMsg)
Proxy Server registration status :
-----------------------------------------------------------------------------------
E.164 UserName Password Port Status
-----------------------------------------------------------------------------------
T addpac 117133 0/ 0 Failed
SIP UA Timer counters
retry counter = 3
SIP UA Timer values
tretry (sip retry timer) = 500 msec.
tinterval (sip retry max interval timer) = 4 sec.
treg (sip register timer) = 60 sec.
tregtry (sip register retry timer) = 20 sec.
texpires (sip invite expire timer) = 180 sec.
tsipping (sip ping timer) = 45 sec.
tsrv (sip srv retry timer) = 60 sec.
thppolling (sip higher priority polling timer) = 30 sec.
SIP UA Session Timer value
Min-SE = 1800 sec.
Session-Expires = 1800 sec.
SIP DNS SRV Query : Disable
SIP Called-Party-Number : from URL
SIP Call Transfer Mode : Basic
SIP Media Channel Start Mode : Default
SIP Reliable Provisional Response Option : Supported with value <100rel>
SIP Response Option : default
SIP Local Domain : NULL
SIP Special Char : NULL
SIP Routing Method of Incoming Call : Default
SIP Remote-Party-ID : Enabled
SIP Local Host Name : No
SIP Conference Server Info
Name (ID) = NULL
Related Voip Tag = -1
SIP NAT Info
PING = Disabled
Required = NULL
SIP Session Refresh Method = INVITE
SIP Keep Authentication information on registration = Yes
SIP Message Parameter Translation TRUE
SIP Fault Torelence = TRUE
SIP Higher Priority Polling = FALSE
SIP Force-Forwarding Info
SIP Hook-Flash Event(INFO) Ignore = FALSE
SIP Time Sync With REGISTER Msg = FALSE
IP-IPO = 192.168.10.13.
Буду блаодарен за любой совет.
Заранее спасибо за помощь.