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VG-204 SIP - PSTN outbound not happening from CUCM

dhananjeyan_t
Level 1
Level 1
I have VG204 configured as SIP , SIP trunk is established to CUCM 7.x , Outbound to PSTN calls is not happening but Extension dialing  within CUCM DB is happening. Inbound from PSTN works.

.

Call manager is not forwarding the calls initiated from vg204 extn to PSTN gateway .
Pls assist if i am missing something on vg204 or CUCM for outbound calls to happen .
, the error log on cucm  is

SIP/2.0 404 Not Found

Reason: Q.850;cause=1


version 15.1

no service pad

service tcp-keepalives-in

service tcp-keepalives-out

service timestamps debug datetime msec localtime show-timezone

service timestamps log datetime msec localtime show-timezone

service password-encryption

no service dhcp

!

!

boot-start-marker

boot system flash:vg20x-advipservicesk9-mz.151-3.T3.bin

boot system flash:

boot-end-marker

!

!

aaa session-id common

clock timezone GMT 0 0

crypto pki token default removal timeout 0

!

!

no ip source-route

!

!

!

!

!

ip cef

no ip bootp server

no ip domain lookup

no ipv6 cef

!        

!

!

!

!

voice call send-alert

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 h323

allow-connections sip to sip

fax protocol pass-through g711alaw

modem passthrough nse codec g711alaw

sip

  bind control source-interface FastEthernet0/0

  bind media source-interface FastEthernet0/0

  redirect contact order best-match

!

voice class codec 2

codec preference 4 g711alaw

codec preference 5 g711ulaw

!

voice class sip-profiles 1000

request ANY sip-header User-Agent modify "User-Agent: Cisco-SIPGateway.*" "User-Agent: Cisco/v11 r1/alg01"

request ANY sip-header Server modify "Server: Cisco-SIPGateway.*" "Server: Cisco/v11 r1/alg01"

response ANY sip-header Server modify "Server: Cisco-SIPGateway.*" "Server: Cisco/v11 r1/h2p-vg-alg01"

response ANY sip-header User-Agent modify "User-Agent: Cisco-SIPGateway.*" "User-Agent: Cisco/v11 r1/alg01"

response ANY sip-header Remote-Party-ID remove

request ANY sdp-header Attribute modify "a=rtr" "a=X-nortr"

response ANY sdp-header Attribute modify "a=rtr" "a=X-nortr"

!

!

!

voice vad-time 1000

!

!

voice-card 0

!

!

application

package callfeature

  param long-dur-disc-cause 21

  param long-dur-duration 1440

  param long-dur-action disconnect

  param long-dur-call-mon enable

!

service dsapp

  param callWaiting FALSE

  param blind-xfer-wait-time 2

  param callTransfer TRUE

!

global

  service default dsapp

!

!

archive

log config

  hidekeys

dial-control-mib retain-timer 1440

dial-control-mib max-size 500

!

!

ip tftp source-interface FastEthernet0/0

!

!

!

!

!

!

!

interface FastEthernet0/0

ip address 10.10.10.13 255.255.255.224

ip access-group al_trusted_SIP in

no ip redirects

no ip unreachables

no ip proxy-arp

ip flow ingress

load-interval 60

speed 100

full-duplex

arp timeout 420

!

control-plane

!

!

voice-port 0/0

no snmp trap link-status

timeouts interdigit 4

description h2p/01

station-id name Analog phone

station-id number 26999

caller-id enable

!

voice-port 0/1

no snmp trap link-status

timeouts interdigit 4

description h2p/02

station-id name Analog phone

station-id number 26998

caller-id enable

!

voice-port 0/2

no snmp trap link-status

timeouts interdigit 4

description h2p/03

station-id name Analog phone

station-id number 26997

caller-id enable

!

voice-port 0/3

no snmp trap link-status

timeouts interdigit 4

description h2p/04

station-id name Analog phone

station-id number 26996

caller-id enable

!

!

!

mgcp profile default

!

!

dial-peer voice 1100 voip

permission orig

description Incoming from IP network

huntstop

session protocol sipv2

incoming called-number .

voice-class codec 2 

voice-class sip profiles 1000

dtmf-relay rtp-nte

!

dial-peer voice 1210 voip

description SIP registrar server

huntstop

preference 4

destination-pattern .T

session protocol sipv2

session target ipv4:10.10.10.15

voice-class codec 2 

voice-class sip profiles 1000

dtmf-relay sip-notify

!

dial-peer voice 99900 pots

description Analog phone

huntstop

preference 4

destination-pattern 26999

progress_ind alert strip

port 0/0

!

dial-peer voice 99901 pots

description Analog phone

huntstop

preference 4

destination-pattern 26998

progress_ind alert strip

port 0/1

!

dial-peer voice 99902 pots

description Analog phone

huntstop

preference 4

destination-pattern 26997

progress_ind alert strip

port 0/2

!

dial-peer voice 99903 pots

description Analog phone

huntstop

preference 4

destination-pattern 26996

progress_ind alert strip

no digit-strip

port 0/3

!

!

dial-peer hunt 2

no dial-peer outbound status-check pots

gateway

media-inactivity-criteria all

timer receive-rtcp 5

timer receive-rtp 1200

!

sip-ua

retry invite 3

registrar 1 ipv4:10.10.10.15 expires 3600

g729-annexb override

!

!

1 Reply 1

dhananjeyan_t
Level 1
Level 1

Can someone assist me pls.