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Busy tone on Incoming calls on SIP based PRI lines

Deepak Kumar
VIP Alumni
VIP Alumni

I am getting busy tone on incoming calls on SIP-based PRI lines. All internal calls are working fine.

Can you help me on CME (SRST) configuration.

My Configuration as below:

SRST#sho running-config
Building configuration...


Current configuration : 8850 bytes
!
!
version 15.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname xxxxx
!
boot-start-marker
boot-end-marker
!
!
card type e1 0 1
enable secret 5 
!
aaa new-model
!
!
aaa authentication login CISCO group radius local
aaa authentication enable default enable
!
aaa session-id common
clock timezone AbuDhab 4 0
network-clock-participate wic 1
!
ip dhcp excluded-address 
ip dhcp excluded-address 
!
ip dhcp pool VOICE
import all
network 
default-router 
option 150 ip 
lease 30
!
ip multicast-routing
ip cef
no ipv6 cef
multilink bundle-name authenticated
!
isdn switch-type primary-net5
!
!
trunk group AUH_FXO
hunt-scheme round-robin
!
cts logging verbose
!
!
voice-card 0
!
voice service voip
no ip address trusted authenticate
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
bind control source-interface GigabitEthernet0/0.60
bind media source-interface GigabitEthernet0/0.60
registrar server expires max 600 min 60
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
!
voice iec syslog
voice register global
mode srst
max-dn 50
max-pool 50
!
voice register pool 1
id network 192.168.50.128 mask 255.255.255.128
no digit collect kpml
dtmf-relay rtp-nte
codec g711ulaw
!
!
voice translation-rule 1
rule 1 /^5\(.*\)/ /022035\1/ type any national plan any isdn
rule 2 /^9\(.*\)/ /\1/ type any national plan any isdn
!
voice translation-rule 2
rule 2 /[0-9]+/ /9&/
!
voice translation-rule 3
rule 1 /^2035\(.*\)/ /5\1/
!
!
voice translation-profile FROM_ISP
translate calling 1
!
voice translation-profile FROM_PSTN
translate calling 2
!
voice translation-profile TO_PSTN
translate calling 1
!
license udi pid CISCO2921/K9 sn FCZ193661MR
hw-module pvdm 0/0
!
!
redundancy
!
!
controller E1 0/1/0
framing NO-CRC4
pri-group timeslots 1-10,16
!

interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
!
interface GigabitEthernet0/0.10
encapsulation dot1Q 10
ip address 192.168.22.5 255.255.255.248
ip pim sparse-mode
ip igmp join-group 239.0.1.2
!
interface GigabitEthernet0/0.60
description #vOICE vLAN#
encapsulation dot1Q 60
ip address 192.168.50.130 255.255.255.128
ip pim sparse-mode
ip igmp join-group 239.0.1.2
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.50.130
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
!
interface Serial0/1/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip pim rp-address 192.168.50.130 PR-GROUP
ip route 0.0.0.0 0.0.0.0 192.168.50.129
!
ip access-list standard PR-GROUP
permit 239.0.1.2
!
control-plane
!
!
voice-port 0/0/0
trunk-group AUH_FXO
translation-profile incoming FROM_ISP
cptone AE
timeouts call-disconnect 5
timeouts wait-release 5
timing guard-out 1000
connection plar 5999
caller-id enable
!
voice-port 0/0/1
trunk-group AUH_FXO
translation-profile incoming FROM_ISP
cptone AE
timeouts call-disconnect 5
timeouts wait-release 5
timing guard-out 1000
connection plar 5999
caller-id enable
!
voice-port 0/0/2
trunk-group AUH_FXO
translation-profile incoming FROM_ISP
cptone AE
timeouts call-disconnect 5
timeouts wait-release 5
timing guard-out 1000
connection plar 5999
caller-id enable
!
voice-port 0/0/3
trunk-group AUH_FXO
translation-profile incoming FROM_ISP
cptone AE
timeouts call-disconnect 5
timeouts wait-release 5
timing guard-out 1000
connection plar 5999
caller-id enable
!
voice-port 0/1/0:15
translation-profile incoming FROM_PSTN
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!

dial-peer voice 1 pots
incoming called-number .
!
dial-peer voice 1000 voip
destination-pattern 5333
session target ipv4:10.10.3.250
incoming called-number .
voice-class codec 1
!
dial-peer voice 1001 voip
destination-pattern 53..
session target ipv4:10.10.3.250
voice-class codec 1
!
dial-peer voice 201 voip
preference 2
destination-pattern 5...
session target ipv4:10.10.3.250
incoming called-number .
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 101 pots
description ****LOCAL CALLS****
translation-profile outgoing did
destination-pattern 9[2-9]......$
direct-inward-dial
port 0/1/0:15
!
dial-peer voice 111 pots
description ****LOCAL CALLS****
translation-profile outgoing did
destination-pattern [2-9]......$
direct-inward-dial
!
dial-peer voice 102 pots
description ****NATIONAL CALLS****
translation-profile outgoing did
destination-pattern 90[2-4,6,7,9].......$
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
prefix 0
!
dial-peer voice 112 pots
description ****NATIONAL CALLS****
translation-profile outgoing did
destination-pattern 0[2-4,6,7,9].......$
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
prefix 0
!
dial-peer voice 103 pots
description ****MOBILE CALLS****
translation-profile outgoing did
destination-pattern 905[0,2,5,6,4,8].......$
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
prefix 05
!
dial-peer voice 113 pots
description ****MOBILE CALLS****
translation-profile outgoing did
destination-pattern 05[0,2,5,6,4,8].......$
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
prefix 05
!
dial-peer voice 104 pots
description ****INTERNATIONAL CALLS****
translation-profile outgoing did
destination-pattern 900T
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
prefix 00
!
dial-peer voice 114 pots
description ****INTERNATIONAL CALLS****
translation-profile outgoing did
destination-pattern 00T
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
prefix 00
!
dial-peer voice 105 pots
description ****TOLLFREE CALLS****
translation-profile outgoing did
destination-pattern 9[6,8]00T
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
!
dial-peer voice 115 pots
description ****TOLLFREE CALLS****
translation-profile outgoing did
destination-pattern [6,8]00T
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
!
dial-peer voice 106 pots
description ****EMERGENCY CALLS****
translation-profile outgoing did
destination-pattern 9[1-9]..$
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
!
dial-peer voice 116 pots
description ****EMERGENCY CALLS****
translation-profile outgoing did
destination-pattern [1-9]..$
progress_ind alert enable 8
direct-inward-dial
port 0/1/0:15
!
dial-peer voice 120 pots
destination-pattern 0[1236789].......
incoming called-number .
direct-inward-dial
port 0/1/0:15
forward-digits 9
!
dial-peer voice 130 voip
destination-pattern 8877
session target ipv4:10.10.3.250
incoming called-number .
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 202 voip
preference 1
destination-pattern 5...
session target ipv4:10.10.3.251
incoming called-number .
voice-class codec 1
dtmf-relay h245-alphanumeric
no vad
!
!
sip-ua
!
gatekeeper
no shutdown
!
!
call-manager-fallback
secondary-dialtone 9
max-conferences 8 gain -6
transfer-system full-consult
timeouts interdigit 5
ip source-address 192.168.50.130 port 2000
max-ephones 50
max-dn 50 dual-line
system message primary O_LOCAL
system message secondary O_LOCAL
transfer-pattern .T
keepalive 20
call-forward pattern .T
call-forward busy 5999
call-forward noan 5999 timeout 45
time-zone 35
date-format dd-mm-yy
!
!
ntp server 10.10.3.1 prefer
!
end

Please guide.

Thanks 

Deepak Kumar

Regards,
Deepak Kumar,
Don't forget to vote and accept the solution if this comment will help you!
2 Replies 2

nathanjgrace
Level 1
Level 1

Greetings, not sure what you mean by SIP-based PRI lines.  We see SIP and PRI config in the router- what's the handoff from the carrier exactly?  Also, was this working at some point and has anything changed recently?

What digits are you receiving from the carrier and what are the internal extensions?

If you are relying on SIP signaling from the carrier, enable debug ccsip messages and provide the debug showing a failed inbound call.  If the signaling comes in the PRI, use debug isdn q931 and provide the output.

Thanks

-Nathan

In addition to Nathan's suggestion, can you also provide the call flow for the inbound call from Telco to the IP Phone.

Thanks