Всем,привет!
Есть проблема со связкой 2-х IP-фонов GXV-3000 через AP1100f. Задача стоит такая: необходимо связать 2 ip-фона через SIP, но без регистрации на SIP сервере. Возникает вопрос: возможно ли из AP1100f сделать SIP-сервер?
Конфиг:
Код:
Using 2254 out of 130868 bytes
!
version 8.30U
!
hostname AP1100F
!
!
no bridge spanning-tree
!
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 192.168.0.11 255.255.255.0
!
interface ether1.0
no ip address
!
snmp name AP1100F
snmp enable-trap dn-register 300 forcely-block
!
no arp reset
!
route 0.0.0.0 0.0.0.0 192.168.0.254
!
logging event all
logging on
!
ntp server 81.22.200.3
!
!
!
!
!
! 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
announcement language russian
!
!
! Voice port configuration.
!
! FXO
voice-port 0/0
ring detect-timeout 70
no caller-id enable
!
!
! FXO
voice-port 0/1
no caller-id enable
!
!
! FXO
voice-port 0/2
no caller-id enable
!
!
! FXO
voice-port 0/3
no caller-id enable
!
!
! FXO
voice-port 1/0
no caller-id enable
!
!
! FXO
voice-port 1/1
no caller-id enable
!
!
! FXO
voice-port 1/2
no caller-id enable
!
!
! FXO
voice-port 1/3
no caller-id enable
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 11 pots
destination-pattern 102
port 0/0
!
!
!
! Voip peer configuration.
!
dial-peer voice 104 voip
destination-pattern 104
session target 192.168.0.18 5060
session protocol sip
answer-address 104
voice-class codec 1
no vad
dtmf-relay rtp-2833
!
dial-peer voice 200 voip
destination-pattern 200
session target 192.168.0.17 5060
session protocol sip
voice-class codec 1
no vad
dtmf-relay rtp-2833
!
!
!
!
!
!
gatekeeper
no shutdown
!
!
! Gateway configuration.
!
gateway
h323-id vo
no ignore-msg-from-other-gk
!
!
! Codec classes configuration.
!
voice class codec 1
codec preference 1 g729
codec preference 4 g711alaw
codec preference 5 g711ulaw
!
!
!
! SIP UA configuration.
!
sip-ua
user-register
call-transfer-mode attended
response alert
!
!
! MGCP configuration.
!
mgcp
codec g729
vad
!
!
! Tones
!
!
!
!
Звонки доходят до AP1100f, но дальше по пирам не идут. Почему это происходит,непонятно.