Откопал железку AP100.
Прошу подскажите ЧЯДНТ?
Шлюз стоит за роутером CISCO 2911, получает статикой IP на Fi 0/0
подключено одно устройство - Panasonic (какая-то радио трубка). Первую минуту все работает нормально, звонки проходят и туда и обратно, работает перевод, в общем все как положено.
конфиг:
Код:
AP100B# sh running-config
Building configuration...
Current configuration:
!
version 8.41.089
!
hostname AP100B
clock timezone Samara 4
!
username root password router administrator
!
!
script ntpdate default
server ip 0.ru.pool.ntp.org
server ip 1.ru.pool.ntp.org
!
interface Loopback0
ip address 127.0.0.1 255.0.0.0
!
interface FastEthernet0/0
ip address 10.33.10.22 255.255.255.0
ip nat outside
speed auto
no qos-control
!
interface FastEthernet0/1
ip address 192.168.10.1 255.255.255.0
ip nat inside
speed auto
no qos-control
!
ip route 0.0.0.0 0.0.0.0 10.33.10.1
!
access-list 100 permit ip 192.168.10.0 0.0.0.255 any
!
!
ip nat inside source list 100 interface FastEthernet0/0 overload
!
!
snmp name AP100B_G2
!
ip tcp keep-alive count 5
ip tcp keep-alive idle 60
ip tcp keep-alive interval 5
ftp server
http server
!
!
dns name-server 8.8.8.8
dns name-server 10.33.8.100
logging event 0-emergency
!
!
!
!
! VoIP configuration.
!
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 0
fax rate 9600
h323 call start fast
h323 call tunnel enable
busyout monitor gatekeeper
busyout monitor sip-server
no busyout monitor callagent
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
caller-id enable
!
!
! FXS
voice-port 0/1
no caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 100
port 0/0
call-waiting
!
dial-peer voice 1 pots
port 0/1
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
voice-class codec 0
no vad
dtmf-relay rtp-2833
!
!
!
!
!
!
! Gateway configuration.
!
gateway
h323-id voip.10.33.10.22
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
!
!
!
! SIP UA configuration.
!
sip-ua
sip-username 100
sip-password pasword
sip-server xxxx.17.rt.ru
timeout treg 600
srv enable
set-local-domain xxxx.17.rt.ru
register e164
!
!
!
!
! MGCP configuration.
!
mgcp
vad
!
!
! Tones
!
!
!
!
line console
!
line vty
!
!
sms
quota 30
!
end
AP100B#
Через минуту звонки престают проходить
debus voip sip дает следующее:
Код:
Sending SIP PDU to ( 357298.17.rt.ru:5060 ) from 5060
REGISTER sip:357298.17.rt.ru SIP/2.0
Via: SIP/2.0/UDP 10.33.10.22:5060;branch=z9hG4bKe55a0d16a41
From: <sip:100@xxxx.17.rt.ru>;tag=e55a0d16a4
To: sip:100@xxxx.17.rt.ru
Call-ID: e554ba5a-b71d-0ddd-8016-0002a4084158@10.33.10.22
CSeq: 1 REGISTER
Date: Tue, 27 Mar 2059 14:27:49 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:100@10.33.10.22>;expires=600
Expires: 600
Content-Length: 0
Max-Forwards: 70
Sending SIP PDU to ( 357298.17.rt.ru:5060 ) from 5060
REGISTER sip:357298.17.rt.ru SIP/2.0
Via: SIP/2.0/UDP 10.33.10.22:5060;branch=z9hG4bKe55a0d16a41
From: <sip:100@xxxx.17.rt.ru>;tag=e55a0d16a4
To: sip:100@xxxx.17.rt.ru
Call-ID: e554ba5a-b71d-0ddd-8016-0002a4084158@10.33.10.22
CSeq: 1 REGISTER
Date: Tue, 27 Mar 2059 14:27:49 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:100@10.33.10.22>;expires=600
Expires: 600
Content-Length: 0
Max-Forwards: 70
Пробовал утсановить параметр keep-alive
Код:
ip tcp keep-alive count 5
ip tcp keep-alive idle 60
ip tcp keep-alive interval 5
к сожлению не помогло(