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CUCM_TO_CUBE_ITSP Provider - SIP/2.0 404 Not Found

ciscoMFNAO
Level 1
Level 1

Hi all,

We are configuring a new SIP Trunk to our new ITSP . Topology looks like: 

CUCM---SIP TRUNK - CUBE - SIP TRUNK - ITSP

 

Calls are being dropped but we can ping ITSP next hop equipment and SIP Servers configured in Dial-peers. Debug shows the error SIP/2.0 404 Not Found.

 

Any ideas what can be causing this issue. Please note CUBE show run and debugs.

For testing We called from 9670 phone extension to 0917775245 PSTN Number

----------------------

 

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2018.02.04 19:25:45 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...


Current configuration : 6218 bytes
!
! Last configuration change at 17:27:18 UTC Sun Feb 4 2018 by cisco
! NVRAM config last updated at 17:11:31 UTC Sun Feb 4 2018 by cisco
! NVRAM config last updated at 17:11:31 UTC Sun Feb 4 2018 by cisco
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname JPTLDAMAN-RTSIP-01
!
boot-start-marker
boot-end-marker
!
!
enable secret 4Y
!
no aaa new-model
!
no ipv6 cef
ip source-route
ip cef
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
!
!
crypto pki token default removal timeout 0
!
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
!
voice service voip
 ip address trusted list
  ipv4 172.29.8.46
  ipv4 0.0.0.0
  ipv4 10.130.2.166
  ipv4 10.130.2.165
 allow-connections h323 to h323
 allow-connections h323 to sip
allow-connections sip to h323
 allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
 rel1xx supported "1XX"
  min-se 90 session-expires 300
!
voice class codec 1
codec preference 1 g711alaw
 codec preference 2 g711ulaw
codec preference 3 g722-64
 codec preference 4 g729r8
codec preference 5 g729br8
 codec preference 6 g722-56
!
voice class sip-profiles 1
request INVITE sip-header Min-SE modify "90" "1800"
 request REINVITE sip-header Min-SE modify "90" "1800"
request INVITE sip-header Session-Expires modify "300" "1800"
 request REINVITE sip-header Session-Expires modify "300" "1800"
!
voice class sip-profiles 2
request INVITE sip-header Min-SE modify "1800" "90"
 request REINVITE sip-header Min-SE modify "1800" "90"
request INVITE sip-header Session-Expires modify "1800" "300"
 request REINVITE sip-header Session-Expires modify "1800" "300"
!
voice class sip-profiles 300
request ANY sdp-header Audio-Attribute modify "inactive" "sendrecv"
 request ANY sdp-header Audio-Attribute modify "sendonly" "sendrecv"
request ANY sdp-header Audio-Attribute modify "recvonly" "sendrecv"
 response ANY sdp-header Audio-Attribute modify "inactive" "sendrecv"
response ANY sdp-header Audio-Attribute modify "sendonly" "sendrecv"
 response ANY sdp-header Audio-Attribute modify "recvonly" "sendrecv"
!
!
!
!
voice translation-rule 1
 rule 1 /^0/ //
rule 2 /^000/ /00/
!
voice translation-rule 2
 rule 1 /^222\(......\)/ /0222\1/
rule 2 /^9\(........\)/ /09\1/
 rule 3 /^244222\(......\)$/ /0222\1/
rule 4 /^2449\(........\)$/ /09\1/
 rule 5 /\(^.+\)/ /000\1/
!
!
voice translation-profile LocalRule
 translate calling 2
!
voice translation-profile SIP-PSTN-OUT
translate called 1
!
!
license udi pid CISCO2911/K9 sn
hw-module pvdm 0/0
!
!
!
username cisco privilege 15 secret 4
!
redundancy
!
!
!
!
!
!
interface Loopback10
no ip address
!
interface Embedded-Service-Engine0/0
 no ip address
shutdown
!
interface GigabitEthernet0/0
 no ip address
shutdown
 duplex auto
speed auto
!
interface GigabitEthernet0/1
 description RTACC08-00-G0/9
ip address 172.29.8.146 255.255.255.0
 no ip redirects
no ip unreachables
 no ip proxy-arp
duplex auto
 speed auto
no mop enabled
!
interface GigabitEthernet0/2
 description SIP To Angola Telecom
ip address 10.130.2.166 255.255.255.252
 no ip redirects
no ip unreachables
 no ip proxy-arp
duplex auto
 speed auto
no mop enabled
!
interface FastEthernet0/0/0
 no ip address
!
interface FastEthernet0/0/1
no ip address
!
interface FastEthernet0/0/2
 no ip address
!
interface FastEthernet0/0/3
no ip address
!
interface Vlan1
 no ip address
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 172.29.8.1
ip route 0.0.0.0 0.0.0.0 10.130.2.165
ip route 10.0.0.0 255.0.0.0 172.29.8.1 name LAN_172
ip route 66.110.114.0 255.255.255.0 10.130.2.165
ip route 172.0.0.0 255.0.0.0 172.29.8.1 name LAN_172
ip route 192.0.0.0 255.0.0.0 172.29.8.1 name LAN_172
!
logging trap debugging
!
!
!
control-plane
!
!
!
!
mgcp profile default
!
!
dial-peer voice 1 voip
redirect ip2ip
 no modem passthrough
session protocol sipv2
 incoming called-number .
voice-class codec 1  
 dtmf-relay sip-notify rtp-nte
fax-relay ecm disable
 no fax-relay sg3-to-g3
fax rate 14400
 no vad
!
dial-peer voice 40 voip
translation-profile outgoing LocalRule
 destination-pattern 2T
no modem passthrough
 session protocol sipv2
session target ipv4:172.29.8.46
 voice-class codec 1  
dtmf-relay sip-notify rtp-nte
 fax-relay ecm disable
no fax-relay sg3-to-g3
 fax rate 14400
no vad
!
dial-peer voice 10 voip
 translation-profile outgoing SIP-PSTN-OUT
destination-pattern 02........
 redirect ip2ip
session protocol sipv2
 session target ipv4:66.110.114.133
voice-class codec 1  
 dtmf-relay sip-notify rtp-nte
fax-relay ecm disable
 fax rate 14400
no vad
!
dial-peer voice 20 voip
 translation-profile outgoing SIP-PSTN-OUT
destination-pattern 000T
 redirect ip2ip
session protocol sipv2
 session target ipv4:66.110.114.133
voice-class codec 1  
 dtmf-relay sip-notify rtp-nte
!
dial-peer voice 50 voip
description #Emergency Calls#
 destination-pattern 01..
session protocol sipv2
 session target ipv4:66.110.114.133
voice-class codec 1  
 dtmf-relay sip-notify rtp-nte
!
dial-peer voice 51 voip
description #Emergency Calls#
 destination-pattern 1..
session protocol sipv2
 session target ipv4:66.110.114.133
voice-class codec 1  
 dtmf-relay sip-notify rtp-nte
!
dial-peer voice 15 voip
translation-profile outgoing SIP-PSTN-OUT
 destination-pattern ^9........
redirect ip2ip
 session protocol sipv2
session target ipv4:66.110.114.133
 dtmf-relay sip-notify rtp-nte
codec g711alaw
!
!
gateway
timer receive-rtp 1200
!
sip-ua
!
!
!
gatekeeper
 shutdown
!
!
!
line con 0
exec-timeout 9 0
 privilege level 0
line aux 0
 exec-timeout 9 0
privilege level 0
line 2
no activation-character
 no exec
transport preferred none
 transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
 stopbits 1
line vty 0 4
 exec-timeout 60 0
privilege level 15
 password cisco
login local
 transport input all
line vty 5 15
 exec-timeout 9 0
privilege level 15
 login
transport input all
!
scheduler allocate 20000 1000
end

JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#
JPTLDAMAN-RTSIP-01#

 

 

-------------------------

 

1 Reply 1

Chris Deren
Hall of Fame
Hall of Fame

What country is this in?

The call is going out as 917775245, is this correct dial string? 404 means the number is incorrect, so perhaps you are sending the incorrect digits to the telco provider.