CUCM Trunk

Top  Previous  Next
Solution Integration > PBXs > BT Call Server > Example Configurations > Trunks Configuration

 

CUCM Trunk

 

Here is the Ad-Hoc procedure to define a Trunk between CUCM and Asterisk:

 

1. Connect to the Call Manager Administration Interface and open the "Device > Trunk" page

2. Select "Add New"

3. Select "SIP Trunk" as Trunk Type

 

Example-Trunk-CUCM_01

 

4. Enter a name, set the "Device Pool" to Default and check "Media Termination Point Required"

 

Example-Trunk-CUCM_02

 

5. Configure the right Calling Search Space and check the "Redirecting Diversion Header Delivery - Inbound" box

 

Example-Trunk-CUCM_03

 

6. Enter the SIP Information of the Trunk => IP address of the Asterisk Server

For a single Asterisk node PBX :

 

Example-Trunk-CUCM_04

 

For dual Asterisk node PBX :

 

SRV DNS records need to be configured :

 

     a) In the DNS server, create a new domain in the main domain

 

Example-Trunk-CUCM_05

 

Example-Trunk-CUCM_06

 

     b) In the newly created domain, add a "Other New Records"  of type SRV

 

Example-Trunk-CUCM_07

 

Example-Trunk-CUCM_08

 

     c) Fill the information with : service _sip, protocol  _tcp, priority 0, weight 0, port 5060 and the IP address of the primary Asterisk

 

Example-Trunk-CUCM_09

 

     d) Create a new SRV record with the information : service _sip, protocol  _tcp, priority 10, weight 0, port 5060 and the IP address of the secondary Asterisk

 

Example-Trunk-CUCM_10

 

     e) Summary of the new SRV records

 

Example-Trunk-CUCM_11

 

     f) In the SIP Trunk on CUCM, check "Destination Address is an SRV" and enter the newly created domain

 

Example-Trunk-CUCM_12

 

 

7. Select "SIP Trunk Security Profile" as "Non Secure", configure the right Calling Search Spaces, use a SIP profile created as shown just below and choose the "RFC 2833" DTMF method

 

Example-Trunk-CUCM_13

 

    The SIP Profile used for the Asterisk trunk is a copy of the Standard SIP Profile with the SIP OPTIONS Ping enabled. See here:

 

Example-Trunk-CUCM_14

 

8. Go to the "Call routing > Route/Hunt > Route Pattern" page

 

9. Select "Add New"

 

10. Set the "Route Pattern" and "Route Partition" accordingly, then select as "Gateway/Route List" the SIP Trunk created above

 

Example-Trunk-CUCM_15

 

11. Repeat step 9 for each Pattern that must be routed to Asterisk.

 

12. The Trunk configuration of Asterisk must have the "remote_hosts" and “identify/match” parameters set to the CUCM IP Address:

 

For a single CUCM node setup (Pub):

 

[CUCM](trunk-defaults)

remote_hosts = IP_address_PBX_Pub:5060

identify/match = IP_address_PBX_Pub

 
For dual (or more) CUCM nodes setup (Pub and Sub):

[CUCM](trunk-defaults)

remote_hosts = IP_address_PBX_Pub:5060,IP_address_PBX_Sub:5060         ; Comma separated IP's (with port) of PBX Pub and Sub

identify/match = IP_address_PBX_Pub,IP_address_PBX_Sub                 ; Comma separated IP's of PBX Pub and Sub