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Dedicated FXO port to line button on IP phone
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12-11-2014 09:15 PM - edited 03-17-2019 01:18 AM
I need some help regarding the topic mentioned. Let me give a brief overview of my issue.
My company has an IP Telephony setup in our office complete with a 2811 router and CUCM 8.6.2 with the router installed with 16 FXO ports.
Now my boss wants me to map a certain FXO trunk line to his extension so any incoming/outgoing call would be to/from his IP Phone. Now I can do something about the incoming calls, redirecting them to his extension using the 'connection plar' but when it comes to outgoing calls, I cannot do this (or don't know) unless I give that voice port a different destination pattern and creating new route pattern for it.
Summarizing it all, he want another DN on his 7962 IP Phone which when picked up directly gives him the external FXO PSTN dial tone and when that PSTN number called will land the call on his DN.
please help me in step by step in voice gateway and cucm menu.
Thanks,
Ali
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12-11-2014 11:40 PM
My thoughts on this, would be to put the FXO port in a new/different partition to all the other FXO ports. Then create a new CSS, add this FXO port's partition to it. Create a new DN and give it access to this new CSS. Then put this new DN as a line button on the phone.
That way, that DN can access the FXO line.
GTG
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12-12-2014 01:42 AM
my main problem is i dont know how can i put the FXO port in a new/different partition to all the other FXO ports.
what are the commands that i have to enter on voice gateway?
thanks.
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12-12-2014 03:51 AM
Hi,
what kind of voice gateway you have ? H323 or MGCP
if it is H323
configure connection plar opx DN under FXO configuration in gateway.
configure different route pattern/calling search space for outbound. associate new CSS with DN.
when you configure the new route pattern, manipulate the called number by adding a prefix 99 for example.
in the gateway, configure a dialpeer with destination pattern 99... ,associate this dialpeer with fxo.
voice translation-rule 1
rule 1 /^99\(.*\)/ /\1/
voice translation-profile remove-99
translate called 1
dial-peer voice 10 pots
destination-pattern 99T
translation-profile outgoing remove-99
forward-digits all
port 0/0/0
in MGCP case.
configure attendant DN in FXO configuration.
configure new RG with this FXO port. create new Route pattern and new CSS for this DN
HTH
Anas
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12-13-2014 12:05 AM
Sorry! I don't have both of them I have installed SIP Trunk with this configuration. My mind goes blank and I don't know what can I do.
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname VoiceGW
!
boot-start-marker
boot bootstrap flash:c2800nm-adventerprisek9-mz.124-15.T3.bin
boot system flash:c2800nm-adventerprisek9-mz.124-15.T3.bin
boot-end-marker
!
logging buffered 51200 warnings
!
no aaa new-model
clock timezone IRST 3 30
!
!
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.100.1 192.168.100.10
!
ip dhcp pool Defaul-Pool
network 192.168.100.0 255.255.255.0
option 150 ip 192.168.100.2
default-router 192.168.100.1
!
!
ip domain name yourdomain.com
ip auth-proxy max-nodata-conns 3
ip admission max-nodata-conns 3
!
multilink bundle-name authenticated
!
!
!
trunk group PSTN
!
voice-card 0
no dspfarm
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
no h225 timeout keepalive
sip
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
voice class h323 1
h225 timeout tcp establish 2
!
voice class custom-cptone IR
dualtone busy
frequency 425 425
cadence 500 500
dualtone ringback
frequency 425 425
cadence 1000 4000
dualtone disconnect
frequency 425 425
cadence 250 250
!
!
!
!
!
crypto pki trustpoint TP-self-signed-37646987
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-37646987
revocation-check none
rsakeypair TP-self-signed-37646987
!
!
crypto pki certificate chain TP-self-signed-37646987
certificate self-signed 01 nvram:IOS-Self-Sig#1.cer
!
!
username admin privilege 15 secret 5 $1$FHPY$P7BBq/euL8Pj9Ue8uIQ2R/
username cisco privilege 15 secret 5 $1$jKQZ$kVpwV5BcnR14CQr/Sat3h1
archive
log config
hidekeys
!
!
!
!
!
!
!
!
interface FastEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-FE 0/0$
ip address 192.168.100.1 255.255.255.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
!
!
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
!
control-plane
!
!
!
voice-port 0/0/0
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone IR
no battery-reversal
input gain 6
output attenuation 2
echo-cancel coverage 24
no vad
timeouts call-disconnect 2
timeouts wait-release 3
timing sup-disconnect 50
connection plar opx 1000
bearer-cap Speech
caller-id enable
!
voice-port 0/0/1
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone IR
no battery-reversal
input gain 6
output attenuation 2
echo-cancel coverage 24
no vad
timeouts call-disconnect 2
timeouts wait-release 3
timing sup-disconnect 50
connection plar opx 1000 -----> Secretary
bearer-cap Speech
caller-id enable
!
voice-port 0/0/2
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone IR
no battery-reversal
input gain 6
output attenuation 2
echo-cancel coverage 24
no vad
timeouts call-disconnect 2
timeouts wait-release 3
timing sup-disconnect 50
connection plar opx 1000
bearer-cap Speech
caller-id enable
!
voice-port 0/0/3
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone IR
no battery-reversal
input gain 6
output attenuation 2
echo-cancel coverage 24
no vad
timeouts call-disconnect 2
timeouts wait-release 3
timing sup-disconnect 50
connection plar opx 1000
bearer-cap Speech
caller-id enable
!
voice-port 0/1/0
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone IR
no battery-reversal
input gain 6
output attenuation 2
echo-cancel coverage 24
no vad
timeouts call-disconnect 2
timeouts wait-release 3
timing sup-disconnect 50
connection plar opx 1000
bearer-cap Speech
caller-id enable
!
voice-port 0/1/1
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone IR
no battery-reversal
input gain 6
output attenuation 2
echo-cancel coverage 24
no vad
timeouts call-disconnect 2
timeouts wait-release 3
timing sup-disconnect 50
connection plar opx 1000
bearer-cap Speech
caller-id enable
!
voice-port 0/1/2
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone IR
no battery-reversal
input gain 6
output attenuation 2
echo-cancel coverage 24
no vad
timeouts call-disconnect 2
timeouts wait-release 3
timing sup-disconnect 50
connection plar opx 1000
bearer-cap Speech
caller-id enable
!
voice-port 0/1/3
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone IR
no battery-reversal
input gain 6
output attenuation 2
echo-cancel coverage 24
no vad
timeouts call-disconnect 2
timeouts wait-release 3
timing sup-disconnect 50
connection plar opx 1000
bearer-cap Speech
caller-id enable
!
voice-port 0/2/0
supervisory disconnect dualtone mid-call
supervisory answer dualtone sensitivity high
supervisory custom-cptone IR
no battery-reversal
input gain 6
output attenuation 2
echo-cancel coverage 24
no vad
timeouts call-disconnect 2
timeouts wait-release 3
timing sup-disconnect 50
connection plar opx 1000
bearer-cap Speech
caller-id enable
!
voice-port 0/2/1
no battery-reversal
no vad
timeouts interdigit 5
timeouts call-disconnect 1
timeouts wait-release 3
connection plar opx 300 -----> this is my DN that i configured in CUCM and CSS
caller-id enable
!
voice-port 0/2/2
shutdown
!
voice-port 0/2/3
shutdown
!
voice-port 0/3/0
description FAX_SRV
station-id number 7010
caller-id enable
!
voice-port 0/3/1
shutdown
!
voice-port 0/3/2
shutdown
!
voice-port 0/3/3
shutdown
!
!
!
!
dial-peer cor custom
name external
!
!
dial-peer cor list external
member external
!
!
dial-peer voice 1 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/0/0
!
dial-peer voice 2 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/0/1
!
dial-peer voice 3 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/0/2
!
dial-peer voice 4 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/0/3
!
dial-peer voice 5 pots
description incoming-PSTN
incoming called-number .
port 0/0/0
!
dial-peer voice 6 pots
description incoming-PSTN
incoming called-number .
port 0/0/1
!
dial-peer voice 7 pots
description incoming-PSTN
incoming called-number .
port 0/0/2
!
dial-peer voice 8 pots
description incoming-PSTN
incoming called-number .
port 0/0/3
!
dial-peer voice 9 voip
description Outgoing-CUCM
destination-pattern 1...
session protocol sipv2
session target ipv4:192.168.100.2:5060
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 10 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/1/0
!
dial-peer voice 11 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/1/1
!
dial-peer voice 12 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/1/2
!
dial-peer voice 13 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/1/3
!
dial-peer voice 14 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/2/0
!
dial-peer voice 15 pots
description Outgoing-PSTN
destination-pattern 9.T
port 0/2/1
!
dial-peer voice 16 pots
description incoming-PSTN
incoming called-number .
port 0/1/0
!
dial-peer voice 17 pots
description incoming-PSTN
incoming called-number .
port 0/1/1
!
dial-peer voice 18 pots
description incoming-PSTN
incoming called-number .
port 0/1/2
!
dial-peer voice 19 pots
description incoming-PSTN
incoming called-number .
port 0/1/3
!
dial-peer voice 20 pots
description incoming-PSTN
incoming called-number .
port 0/2/0
!
dial-peer voice 21 pots
description incoming-PSTN
incoming called-number .
port 0/2/1
!
dial-peer voice 7010 pots
description FAX_SRV
destination-pattern 7010
port 0/3/0
!
dial-peer voice 300 voip
destination-pattern 300
session target ipv4:192.168.100.2
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
!
gateway
timer receive-rtp 1200
!
sip-ua
registrar ipv4:192.168.100.2:5060 expires 3600
sip-server ipv4:192.168.100.2:5060
!
!
!
line con 0
login local
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
!
scheduler allocate 20000 1000
ntp master
ntp server 192.168.100.1
!
end
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12-13-2014 12:38 AM
Hi,
also with SIP trunk you can use the same steps like H323.
the idea is make the patterns dialed by your specific line is different from the other patterns.
In that way you can match specific dial-peer which is pointing to port 0/2/1
One more thing, you dial-peer voice 300 voip is H323 not SIP
you need to configure it as SIP.
by adding to below commands under dial-peer
sip protocol sipv2
dtmf-relay sip-notify
HTH
Anas
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12-13-2014 01:19 PM
I'm Really soryy but I have changed my configuration to the following but when i press the button i have to press 0 for two time. you know i dont want to press any access digit for instance 9 or any other code. I'm going to press the button and start to dialing out the digits.
access fxo line from ip phone button directly without any access code!!!
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12-13-2014 02:26 PM
Hi.
I'll try to give you an example of what both Gordon and Anas ( +5 both) are trying to suggest you.
First you have to do some configuration on your CUCM:
Create a new partition Out_from_300
Create a new CSS CSS_For_300 and put the newly created partition and all partitions that extension 300 needs to reach internal extensions.
create a new route pattern.! and put in Out_from_300 partition and put 999 on outgoing prefix field
Add the newly created CSS to directory number 300.
On VG
configure this dialpeer
dial-peer voice 9000 pots
destination-patter 999T
port x/x/x
In this case you will be able to pick up your phone and start dialing without any access code
HTH
Regards
Carlo
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02-08-2017 11:14 PM
Hi Carlo,
Do I need to create translation pattern along with this ? I am in to same scenario.
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