The configuration is basically having CUCM to handle 1XXX extensions and Trixbox to handle 2XXX extensions. Tribox is central IP PBX to handler outgoing and incoming calls.
The main goals for this Part#1 are:
- To be able to make a phone call between IP Phones by dialling their extension numbers (e.g. 1000 -> 1001, 1000 -> 2001, 1000 -> 2000, 2000 -> 1001, etc)
- Dial 9, followed by the numbers, to dial outside world via VOIP
- Dial 0, followed by the numbers, to dial outside world via PSTN
- To be able to receive incoming call from either PSTN or VOIP DID number and rings my IP Phone(s)
Here is the data/voice layout:
The version being used is:
CUCM: 6.0.1.2000-4
Trixbox: 2.8.0.1
SPA3102 Firmware: 5.1.10(GW)
DHCP Setting
Add the following DHCP options point to the CUCM IP address
Options 66
Options 150
Cisco Unified CM Configuration
Enable Services
Go to Cisco Unified Serviceability -> Tools -> Service Activation, enable the following services:
Cisco CallManager
Cisco Tftp
Cisco IP Voice Media Streaming App
Start Services
Go to Cisco Unified Serviceability -> Control Center -> Feature Services, start the following services:
Cisco CallManager
Cisco Tftp
Cisco IP Voice Media Streaming App
Auto Registration
Go to Cisco Unified CM Administration -> System -> Cisco Unified CM
Starting Directory Number*: 1000 (for example)
Ending Directory Number*: 1500 (for example)
Un-tick "Auto-Registration Disabled on this Cisco Unified Communication Manager" checkbox
Cisco Unified CM Group Configuration
Go to Cisco Unified CM Administration -> System -> Cisco Unified CM Group
Create a new Group
Give a name
Tick "Auto-registration Cisco Unified Communications Manager Group" checkbox
Add the CUCM Server to the group member
Phone NTP Reference
Go to Cisco Unified CM Administration -> System -> Phone NTP Reference
Add a new Reference
Assign an IP Address
Mode* = default
Date/Time Group
Go to Cisco Unified CM Administration -> System -> Date/Time Group
Create a new Group
Give a name
Select time Zone, separator, date format and time format
Select NTP References from the configuration above
Device Pool
Go to Cisco Unified CM Administration -> System -> Device Pool
Create a new Device Pool
Give a name
Cisco Unified Communications Manager Group* = #select from the one created above
Region* = default
SRST Reference* = Use Default Gateway
SIP Trunk Security Profile
Go to Cisco Unified CM Administration -> System -> Security Profile -> SIP Trunk Security Profile
Create a new Profile
Give a name
Incoming Transport Type*= TCP_OR_UDP
Outgoing Transport Type*= USER_DATAGRAM_PROTOCOL
Incoming Port*= 5060
Create SIP Trunk
Go to Cisco Unified CM Administration -> Device -> Trunk
Add a new SIP Trunk
Give a name
Device Pool* = #select from the one created above
Route Group
Go to Cisco Unified CM Administration -> Call Routing -> Route/Hunt -> Route Group
Add a new one
Give a name
Add the SIP Trunk created above to the member of the Route Group
Route List
Go to Cisco Unified CM Administration -> Call Routing -> Route/Hunt -> Route List
Add a new one
Give a name
Cisco Unified Communications Manager Group* = #select from the one created above
Route Option = Route this pattern
Tick "Provide Outside Dial Tone" checkbox
Add a new oneRoute Pattern* = 0.!
Gateway/Route List* = #select from the one created above
Route Option = Route this pattern
Trixbox Configuration
Create SIP Trunk to CUCM
Go to PBX -> PBX Settings -> Trunks
Add SIP Trunk
Trunk Name = CUCM
Peer Details:
disallow=all
type=friend
host=
allow=ulaw&alaw
nat=no
canreinvite=yes
qualify=yes
User Context: CUCM-IN
User Settings:
context=from-internal
host=#cucm-address
type=friend
Create SIP Trunk to VOIP Provider
Go to PBX -> PBX Settings -> Trunks
Add SIP Trunk
Outbound Caller ID: "Name"
Dial Rules:
612+NXXXXXXX
04.
0011.
61+13XXXX
61+1800XXXXXX
#Note: Dial rules are created because my VOIP Provider requires the number format to be International format
Trunk Name = voip
Peer Details:
allow=alaw&ulaw&gsm
canredirect=no
canreinvite=no
disallow=all
host=
insecure=very
secret=
type=peer
username=
User Context: User
User Details:
canreinvite=no
context=fromtrunk
fromuser=
qualify=no
secret=
type=user
username=
Registration String:
Create SIP Trunk to PSTN
Go to PBX -> PBX Settings -> Trunks
Add SIP Trunk
Outbound Caller ID: "DID"
Maximum Channels: 1
Trunk Name: pstn
Peer Details:
disallow=all
allow=ulaw
canreinvite=no
context=fromtrunk
dtmfmode=rfc2833
host=
incominglimit=1
nat=never
port=5061
qualify=yes
secret=
type=friend
username=pstn
Create Outbound Routes
#There will be 3 route patterns:
Dial 9, to go to VOIP
Dial 0, to go to PSTN
Dial 1XXX to go to SIP Phones registered with CUCM
Go to PBX -> PBX Settings -> Outbound Routes
Add Route
Name: VOIP
Dial Patterns: 9.
Trunk Sequence: SIP/Pennytel
Add Route
Name: PSTN
Dial Patterns: 0.
Trunk Sequence: SIP/pstn
Add Route
Name: CUCM
Dial Patterns: 1XXX
Trunk Sequence: SIP/CUCM
Allowing Incoming SIP Calls
Go to PBX -> PBX Settings -> General Settings
Allow Anonymous Inbound SIP Calls: Yes
Create Ring Groups
# This will allow incoming call to ring both my Cisco IP Phone (ext 1000) and my analog phone which connected to my SPA3102 (ext 2000)
Go to PBX -> PBX Settings -> Ring Groups
Give description
Ring strategy: ringall
Extension List:
2000
1000#
Create Inbound Routes
Go to PBX -> PBX Settings -> Inbound Routes
#Make sure to leave the DID and Caller ID number blank - this will accept all incoming routes
Set Destination:
Ring Group: #select the one created above
