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Configure SIP TRUNK on CISCO 2851

livedigital
Level 1
Level 1

Hi I have configured SIP trunk from service provider for the incoming calls so that it lands on to our VXML application . When I am calling , there are no SIP events generated . I am able to ping the SIP server of service provider . Plz help

Configuration is as below 

 

Building configuration...


Current configuration : 2098 bytes
!
! Last configuration change at 11:36:37 UTC Thu Jul 10 2014
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot system flash:c2800nm-adventerprisek9-mz.151-4.M3.bin
boot-end-marker
!
!
!
no aaa new-model
!
no network-clock-participate wic 0
no network-clock-participate wic 1
no network-clock-participate wic 2
no network-clock-participate wic 3
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
!
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
 no ip address trusted authenticate
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to sip
 redirect ip2ip
 sip
  bind control source-interface GigabitEthernet0/0
  bind media source-interface GigabitEthernet0/0
!
voice class codec 1
 codec preference 2 g711ulaw
!
!
!
!
!
voice-card 0
!
!
application
 service first http://203.122.28.180/try.vxml
 !
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO2851 sn FHK1009F1NQ
!
redundancy
!
!
controller E1 0/0/0
!
controller E1 0/1/0
!
controller E1 0/2/0
!
controller E1 0/3/0
!
!
!
!
!
!
!
!
!
interface GigabitEthernet0/0
 ip address 10.50.129.82 255.255.255.252
 duplex auto
 speed auto
!
interface GigabitEthernet0/1
 ip address 203.122.28.181 255.255.255.248
 duplex auto
 speed auto
!
ip forward-protocol nd
ip http server
no ip http secure-server
!
!
ip route 10.0.0.0 255.0.0.0 10.50.129.81
ip route 203.0.0.0 255.0.0.0 203.122.28.177
!
!
!
!
!
!
!
control-plane
!
!
!
!
mgcp profile default
!
!
dial-peer voice 3000 voip
 service first
 session protocol sipv2
 session target ipv4:10.50.129.50
 incoming called-number .
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 no vad
!
!
sip-ua
 authentication username 6671111 password 7 101F5B4A51
 registrar ipv4:10.50.129.50 expires 3600
!
!
!
!
line con 0
line aux 0
line vty 0 4
 login
 transport input all
!
scheduler allocate 20000 1000
end

1 Reply 1

ajay1komban
Level 1
Level 1

 

Dear live,

 

First you should need the connectivity ,then only you can create sip trunk

When you ping to service provider,what is the reply getting ?

Regards,

Ajay Jose k