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Cisco 827 4v sip and h323


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





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

ip name-server


voice rtp send-recv


voice service voip

allow-connections h323 to h323

allow-connections sip to sip



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

no ip redirects

no ip route-cache

no ip mroute-cache

no cdp enable

hold-queue 100 out


interface ATM0

no ip address


no atm ilmi-keepalive

dsl operating-mode auto


ip forward-protocol nd

ip route

ip http server


no cdp run




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

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

session transport udp

dtmf-relay rtp-nte


dial-peer voice 1 pots

answer-address 220000

destination-pattern 220000

port 1





authentication username 220000 password xxxxxx

nat symmetric check-media-src

max-forwards 10

registrar expires 3600


no suspend-resume


line con 0

line vty 0 4

privilege level 15

login local


scheduler max-task-time 5000

sntp server



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Re: Cisco 827 4v sip and h323


I'm not sure of what you are trying to do, but to find out where calls are going, use "show dialplan number XXXX" and "debug voip dialpeer all" with "term mon", to find how calls are routed.

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