Dear.
I have Cisco 827-4V.
Voice-port 1 connection Plar.
Voice port 2 sip-ua to CommunigatePRO (sip-proxy)
My problem.
When i wont make call to PSTN(plar-h323), all call going to sip(CommunigatePRO)
Sip is work fine.
Please tel me, where is mistake or give link to manual.
Just make sure everybody understood my question.
I'm true to make call from port 1 to pstn, or port 2 to sip, when try from plar, plar going to sip.
Sorry for my English.
Thanks Andriy.
-----------------------------
sh running-config
version 12.4
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Cisco-827-4v
!
boot-start-marker
boot-end-marker
!
no aaa new-model
clock timezone London 0
clock summer-time London recurring last Sun Mar 2:00 last Sun Oct 2:0
!
ip name-server xxx.xxx.xxx.xxx
ip name-server xxx.xxx.xxx.xxx
!
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections sip to sip
!
application
service dsapp param
param callWaiting TRUE
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g726r32
codec preference 4 g726r24
codec preference 5 g726r16
codec preference 6 g723r53
codec preference 7 g723ar53
codec preference 8 g723ar63
codec preference 9 g711alaw
codec preference 10 g711ulaw
!
username andriy privilege 15 password xxxxxxxxxx
!
ip ftp username admin
ip ftp password 2003
!
interface Ethernet0
ip address xxx.xxx.xxx.xxx 255.255.255.248
no ip redirects
no ip route-cache
no ip mroute-cache
no cdp enable
hold-queue 100 out
!
interface ATM0
no ip address
shutdown
no atm ilmi-keepalive
dsl operating-mode auto
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.xxx
ip http server
!
no cdp run
!
control-plane
!
voice-port 1
!
voice-port 2
connection plar 256745
!
voice-port 3
!
voice-port 4
!
dial-peer voice 3 voip
destination-pattern [0-9]......+
session target ipv4:xxx.xxx.xxx.xxx
dtmf-relay h245-signal
codec g729br8
!
dial-peer voice 2 pots
destination-pattern 256745
port 2
!
dial-peer voice 220000 voip
service session
destination-pattern 22....
voice-class codec 1
session protocol sipv2
session target dns:xxx.xxx.xxx.xxx
session transport udp
dtmf-relay rtp-nte
!
dial-peer voice 1 pots
answer-address 220000
destination-pattern 220000
port 1
!
gateway
!
sip-ua
authentication username 220000 password xxxxxx
nat symmetric check-media-src
max-forwards 10
registrar dns:xxx.xxx.xxx.xxx:5060 expires 3600
sip-server dns:xxx.xxx.xxx.xxx
no suspend-resume
!
line con 0
line vty 0 4
privilege level 15
login local
!
scheduler max-task-time 5000
sntp server xxx.xxx.xxx.xxx
end
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