Trixbox SIP trunk to Cisco Unified Call Manager 7.X

zsyzhou

Some time ago, I needed to configure an SIP trunk between a Trixbox (Asterisk on Linux) PBX and a Cisco Call Manager PBX. It was pretty hard to find any relevant information on the internet, however eventually I figured out how to do it.
　　Please note that the following configuration reflects a Trixbox PBX configured with phones with extensions of 1XX and the Cisco Unified Call Manager configured with extensions of 3XX.
　　Trixbox Configuration
　　Create an SIP Trunk (Leave settings default unless otherwise specified below)
　　Outgoing Settings
　　Trunk Name: CallManager
　　Peer Details:
　　type=friend
　　qualify=yes
　　nat=no
　　insecure=very
　　host=ip.address.of.CUCM
　　fromdomain=ip.address.of.CUCM
　　dtmf=rfc2833
　　disallow=all
　　context=from-internal
　　canreinvite=no
　　allow=ulaw
　　Incoming Settings
　　USER Context: ip.address.of.CUCM
　　USER Details:
　　type=friend
　　qualify=yes
　　nat=no
　　insecure=very
　　host= ip.address.of.CUCM
　　fromdomain= ip.address.of.CUCM
　　dtmf=rfc2833
　　disallow=all
　　context=from-internal
　　canreinvite=no
　　allow=ulaw
　　Create an Outbound Route to route calls made to 3XX to the Cisco Call Manager
　　Create outbound route “Cisco”. Check the “Intra Company Route”, and inside of the Dial Patterns type in 3XX. Under Trunk Sequence select “CallManager”.
　　This pretty much sums up the amount of configuration required on the Trixbox side of things. Now onto the Cisco stuff.
　　Cisco Unified Call Manager Configuration
　　Create an SIP Trunk
　　Device -> Trunk -> Add New
　　Trunk Type: SIP Trunk
　　Device Protocol: SIP
　　Device Name: TrixboxPBX
　　Call Classification: OnNet
　　Check the “Media Termination Point Required” checkbox (this is to handle transfers, hold music, etc…)
　　Check “Remote-Party-Id”
　　Check “Asserted-Identity”
　　SIP Information
　　Destination Address: IP.address.of.trixbox
　　Uncheck “Destination Address is an SRV”
　　Destination Port: 5060
　　MTP Preferred Originating Code: 711ulaw
　　SIP Trunk Security Profile: Non-Secure SIP Trunk Profile
　　Change the “Non-Secure SIP Trunk Profile” security profile from TCP to UDP
　　System -> Security Profile -> SIP Trunk Security Profile
　　Hit the “Find” button
　　Select “Non Secure SIP Trunk Profile”
　　Incoming Transport Type: TCP+UDP
　　Outgoing Transport Type: UDP
　　Uncheck “Enable Digest Authentication”
　　Incoming Port: 5060
　　Out of the last 6 checkboxes, all should be checked except the First and Last.
　　Create a Route Pattern to route calls from the Cisco Call Manager to Trixbox
　　Call Routing -> Route/Hunt -> Route Pattern
　　Create New
　　Route Pattern: 1XX
　　Gateway/Route List: TrixboxPBX
　　Route Option: Route this pattern
　　Call Classification: OnNet
　　Enable Required Services
　　I’m not too sure which ones are actually required, however the below configuration works great. To get to the CUCM services go to the “Cisco Unified Serviceability” section (Top right of web interface).
　　Enable Services
　　Tools -> Serviceability
　　Enable the following:
　　CM Services
　　Cisco CallManager
　　Cisco Tftp
　　Cisco Messaging Interface
　　Cisco Unified Mobile Voice Access Service
　　Cisco IP Voice Media Streaming App
　　CTI Services
　　Cisco CallManager Attendant Console Server
　　Cisco IP Manager Assistant
　　Cisco WebDialer Web Service
　　Select “Save”, afterwards select “Set to Default”. Please note that it may take some time to bring the services up.
　　It’s always a good idea to restart both the Trixbox PBX and the CUCM PBX.
　　After you have configured the above, configure phones in the 1XX range for the trixbox, configure phones on the CUCM for the 3XX range and they should be able to call each other. Please remember that if you have a PSTN line on your Trixbox you will need to create another route pattern for how to transfer 9XXXXXXXXXX from your CUCM -> Trixbox, then configure the applicable route in Trixbox -> PSTN.
　　Feedback is welcome!