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
Thanks -
Dan
