07-10-2014 01:49 AM - edited 03-16-2019 11:21 PM
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
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! 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
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hostname Router
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boot-start-marker
boot system flash:c2800nm-adventerprisek9-mz.151-4.M3.bin
boot-end-marker
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!
!
no aaa new-model
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no network-clock-participate wic 0
no network-clock-participate wic 1
no network-clock-participate wic 2
no network-clock-participate wic 3
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dot11 syslog
ip source-route
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!
ip cef
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!
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no ipv6 cef
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multilink bundle-name authenticated
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!
!
!
!
!
voice call carrier capacity active
voice rtp send-recv
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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
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!
!
!
!
voice-card 0
!
!
application
service first http://203.122.28.180/try.vxml
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!
crypto pki token default removal timeout 0
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!
!
!
license udi pid CISCO2851 sn FHK1009F1NQ
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redundancy
!
!
controller E1 0/0/0
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controller E1 0/1/0
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controller E1 0/2/0
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controller E1 0/3/0
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!
!
!
!
!
!
!
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interface GigabitEthernet0/0
ip address 10.50.129.82 255.255.255.252
duplex auto
speed auto
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interface GigabitEthernet0/1
ip address 203.122.28.181 255.255.255.248
duplex auto
speed auto
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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
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!
!
!
!
!
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control-plane
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!
!
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mgcp profile default
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!
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
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!
!
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line con 0
line aux 0
line vty 0 4
login
transport input all
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scheduler allocate 20000 1000
end
07-10-2014 10:46 PM
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
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