12-17-2013 04:44 AM - edited 03-16-2019 08:54 PM
Hi Guys,
I have a SIP trunk setup with a 2811 running CME version 7. I can make outbound calls ok but having issues getting the incoming calls working, i have 1 number on my SIP trunk and that is 01133501788 and i want that to ring my Cisco 7960 which is running SIP firmware not SCCP. I have included by config for anyone who can help me, i just want the incoming call to work.
Many Thanks.
Matthew.
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
clock timezone GMT 0
!
dot11 syslog
ip source-route
!
!
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
!
ip dhcp pool DATA_POOL
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
dns-server 188.92.232.50 188.92.232.100
!
ip dhcp pool VOICE_POOL
network 192.168.1.0 255.255.255.0
default-router 192.168.1.1
dns-server 188.92.232.50 188.92.232.100
option 150 ip 192.168.1.1
!
!
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/1.20
bind media source-interface FastEthernet0/1.20
registrar server
!
!
!
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
!
!
!
!
!
!
!
!
!
!
!
!
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0008072514198272
!
voice register dn 1
number 6999
allow watch
name SIP
label SIP
!
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
!
!
voice translation-rule 1
rule 1 /^9\(.*\)/ /\1/
!
voice translation-rule 2
rule 1 /^6...$/ /4143*002/
!
!
voice translation-profile DiscardDigit9
translate calling 2
translate called 1
!
voice translation-profile IncomingSIP
translate calling 1133501788
!
!
voice-card 0
no dspfarm
!
!
!
!
!
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
!
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
!
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
!
!
ip nat inside source list 1 interface FastEthernet0/0 overload
!
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
!
!
!
!
!
!
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
!
control-plane
!
!
!
!
mgcp behavior g729-variants static-pt
!
!
dial-peer cor custom
!
!
!
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing DiscardDigit9
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
!
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
!
!
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
!
!
!
gatekeeper
shutdown
!
!
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
line con 0
line aux 0
line vty 0 4
login
!
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router#
Solved! Go to Solution.
12-17-2013 05:18 AM
Add these....
voice translation-rule 3
rule 1 /^01133501788$/ /6999/
rule 2 /^1133501788$/ /6999/
voice translation-profile IncomingSIP
translate called 3
--------------------------------------------------
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
12-17-2013 05:18 AM
Add these....
voice translation-rule 3
rule 1 /^01133501788$/ /6999/
rule 2 /^1133501788$/ /6999/
voice translation-profile IncomingSIP
translate called 3
--------------------------------------------------
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
12-17-2013 06:54 AM
You my friend are a star! worked straight away, many thanks. Just one more thing, when i make an outgoing call, it always appears as "blocked" on my phone, my sip trunk is set to allow CME to alter outgoing CLI's how would i program the outgoing CLI to 01133501788 also?
The new working config is below with your suggestion, which works!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
clock timezone GMT 0
!
dot11 syslog
ip source-route
!
!
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
!
ip dhcp pool DATA_POOL
network 10.10.10.0 255.255.255.0
default-router 10.10.10.1
dns-server 188.92.232.50 188.92.232.100
!
ip dhcp pool VOICE_POOL
network 192.168.1.0 255.255.255.0
default-router 192.168.1.1
dns-server 188.92.232.50 188.92.232.100
option 150 ip 192.168.1.1
!
!
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server
!
!
!
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
!
!
!
!
!
!
!
!
!
!
!
!
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0015244443466064
!
voice register dn 1
number 6999
allow watch
name SIP
label SIP
!
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
!
!
voice translation-rule 1
rule 1 /^6...$/ /4143*002/
!
voice translation-rule 3
rule 1 /^01133501788$/ /6999/
rule 2 /^1133501788$/ /6999/
!
!
voice translation-profile IncomingSIP
translate called 3
!
voice translation-profile Translatetrunk
translate calling 1
!
!
voice-card 0
no dspfarm
!
!
!
!
!
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
!
!
!
!
!
!
!
!
!
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
!
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
!
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
!
!
ip nat inside source list 1 interface FastEthernet0/0 overload
!
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
!
!
!
!
!
!
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
!
control-plane
!
!
!
!
mgcp behavior g729-variants static-pt
!
!
dial-peer cor custom
!
!
!
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target sip-server
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
!
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing Translatetrunk
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
!
!
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
!
!
!
gatekeeper
shutdown
!
!
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp 7960 Dec 17 2013 14:35:13
!
!
line con 0
line aux 0
line vty 0 4
login
!
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router#
12-17-2013 07:04 AM
voice translation-rule 1
rule 1 /^6...$/ /01133501788/
12-17-2013 07:36 AM
I already have a voice translation rule 1 in place, that strips off the local extension header and replaces it with my SIP trunk header.
6999 is my local extension, that rule replaces 6999 with 4143*002, is there another way to do this?
This is just to manipulate the caller outgoing CLI?
12-17-2013 08:12 AM
Correct me if i am wrong , you want your caller id to be 01133501788 when making outgoing call ?
Now my question is why are you sending 4143*002 as your caller id? Is this something your ITSP asked for? You are already doing authentication through sip-ua.
Sent from Cisco Technical Support iPhone App
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