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Configuring SIP trunk in CME/CUBE

When I first started playing around with SIP trunks and registering it. Had issues and with help, reading and playing around with the configurations. This is what I found. I decided I should share this since I have seen a few posts about configuring SIP trunks. The below configuration is only for credential based SIP trunks. The other SIP trunk is IP address based, where the calls are validated at the SIP providers end is by the Public IP.

  • Please find the configurations steps below if you have multiple direct dials and would like to display each number when

the call is made. e.g. extension 2001 with Direct Dial of 01xxxxxxx2001 calls, he wants that number to be displayed.

If the below mentioned configuration is used, ensure that translation rules are in place, since the SIP provider looks

for a number that is part of the trunk to validate the call.


credentials username xxxxxxxxxxxxx password  xxxxxxxxxxxxxx realm

authentication username xxxxxxxxxxxxxxx password xxxxxxxxxxxxxx realm [SI1]

no remote-party-id                                         [SI2]

retry register 10

timers register 1000


expires 60

voice service voip

ip address trusted list

  ipv4 <add IPs>

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip            [SI3]

supplementary-service h450.12

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none


  registrar server expires max 1500 min 500

  no call service stop

  • Say you ahve only one number in your trunk and all outgoing calls will be sent with the same number.

Under SIP-UA add the following command with the rest of the other commands above. You will not require

a translation rule for outbound since you will be sending a single number.


calling-info pstn-to-sip from number set 01xxxxxxxxx

  • If you ahve Multiple SIP trunks dont panic, just add another credential line (similar to point 1).

I hope this document will be useful to you.

If you have any questions please do not hesitate to comment below and I will be happy to assist you.

If you like the post please rate.

[SI1]This information will be provided by the SIP provider

[SI2]This is done so that the original number is stripped (sometimes in a sip to sip call the extension number is displayed)

[SI3]Allow  different protocol call

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Please post questions in "Discussions", not Documents.

Community Member

Hey Paolo,

Its not a question its a solution if anyone wants to know.

Community Member

Hello Friends,

Does anybody know how to set up a CUBE gateway with CME?


I was reading and it´s strongly recommended to implement CUBE gateway in a telephony Cisco solution for security reasons.

You are partially right. CUBE is recommended when you are in CUCM environment and CUBE sits in between CUCM cluster and service provider.

However In your scenario, CME and CUBE will probably be located in the same box so you don't need to see it as CME and CUBE integration. Just go ahead with your desired configuration and you will be good.  

Community Member


So If there is no CUCM enviroment,it is not necessary to prepare CUBE?

I am still confusing the benefit of CUBE.what is good to have it.


You will mostly see CUBE with CUCM deployment although it is not necessary. CUBE can act as your centralized dial plan and can point multiple cluster to single SIP trunk provider.

- Vivek