# Programming SIP Trunks

## Before You Begin

Ensure you have the following information:

- the IP address or Fully Qualified Domain Name (FQDN) of the SIP peer

- the transport type and the port number for the SIP peer unless it's obtained automatically through DNS

- the FQDN or IP address of the Registrar (if applicable)

- the transport type and the port number for the Registrar unless it's obtained automatically through DNS (if applicable)

- the FQDN or IP address of the outbound proxy server, the transport type, and port number, if an outbound proxy server is being used

- the FQDN or IP address of the external proxy, the transport type, and port number, if an external SIP proxy is being used

- the DID numbers assigned to the MiVoice Business system by the Service Provider

- the prefix that is being added to all incoming calls for a gateway

- the main DID number for registration (if applicable)

- the RTP (voice) stream packet rate that the SIP Service Provider expects

## About Programming SIP Trunks

The licensing for SIP trunks is enabled through the SIP Trunks field in the License and Option Selection form. Each SIP Peer has access to the whole pool of licenses (up to 2000), which can be shared between SIP Peer Profiles. The Maximum Simultaneous Calls field provides the upper limit of how many licenses can be used by a particular SIP Peer Profile. You can also reserve some or all of the assigned licenses for a particular peer (see Minimum Reserved Call Licenses field in the SIP Peer Profile form). Once the Maximum Simultaneous Calls is reached, the call is rejected.

Examples of SIP trunk license configurations for a SIP Peer Profile:

- If you wish to reserve all assigned licenses for a particular peer and not to share any, configure the Minimum Reserved Call Licences field with the same number as the Maximum Simultaneous Calls.

- If you wish to use just shared licenses, set the Minimum Reserved Call Licenses to 0 and the Maximum Simultaneous Calls to the number allocated by your SIP Proxy, SIP Server, or SIP Service Provider.

- If you wish to reserve some and use some shared licenses, configure the Minimum Reserved Call Licenses and the Maximum Simultaneous Calls fields with the required numbers (for example, reserve a minimum of 4 and allow up to 30 calls).

NOTES

- SIP trunks sharing between MiVoice Business systems is not supported.

- Traffic reporting is not supported on SIP Trunks.

## Conditions/Dependencies

- The Class of Service for all devices connected to SIP Trunks must include the option Public Network Access via DPNSS set in the Class of Service Options form.

- If FQDN's are used, ensure that the Domain Name Server (DNS) in the System IP Properties is programmed and the DNS Server is available.

- The route must be deleted from the ARS Routes form before deleting the SIP Peer Profile.

- A SIP Peer Profile must be deleted from the SIP Peer Profile form before deleting the SIP Peer from the Network Elements form.

- Before deleting a SIP Peer Profile, all Outgoing DID Ranges for a SIP Peer Profile must be deleted.

- Before clearing an Index from the DID Ranges for CPN Substitution form, all entries in the Outgoing DID Ranges form using the index must be removed.

- A SIP Peer profile label used by the ARS Routes form or the SIP Peer Profile by Incoming DID form must be removed before removing the SIP Peer Profile.

- If Challenge-based incoming call authentication is required, set the authentication realm string on the remote peer to the network element name of the local network element. If this name is not specified, set the authentication string to the domain name programmed in the System IP Properties form. Some SIP authentication implementations only require the username and password.

- If Challenge-based outgoing call authentication is required, MiVoice Business does not require provisioning of the authentication realm string. Only the username and password are required.

## Programming

Fill out the forms below in the order listed.

1. License and Option Selection form

- Enter the total number of licenses in the SIP Trunks field.

2. Network Elements form

3. Create a network element for the local switch

4. Create a network element for the Outbound Proxy if one exists in your network

5. Create a network element for each SIP peer, gateway, or Service Provider

   NOTE: To add a SIP peer, click Add, select Other as the Type, select the SIP Peer check box, and then program the SIP Peer Specific fields.

3. System IP Ports form

- Change the SIP UDP, TCP, or TLS port number if it is different from the default value.

4. SIP Peer Profile form

- Import Form Data or enter the required information for each SIP peer. The system supports up to 250 SIP Peer Profiles.

5. SIP Peer Profile Assignment by Incoming DID

- To associate a range of telephone numbers assigned by a SIP Service Provider to a particular SIP Peer, enter the required information in this form.

6. Trunk Attributes

Configure at least one of the following (trunk as non-dial in or dial-in) to receive incoming calls:

7. update the Non-Dial-In Trunks Answer Point field for the incoming calls.

8. strip the number of leading digits in Dial-In Trunks Incoming Digit Modification Absorb field

9. add the appropriate number of digits in Dial-In Trunks Incoming Digit Modification Insert field.

7. ARS Routes

ARS continues to be provisioned in the traditional manner by assigning a route to a digit or digit string. Complete the following fields in this form:

7. select SIP Trunk from the pull-down list in Routing Medium.

4. select a SIP Peer Profile label from the SIP Peer Profile pull-down list.

5. enter a Class of Restriction group number in COR Group Number.

6. enter any required digits in Digits Before Outpulsing. (If this field is left blank, digits will be sent out as " Enbloc".)

8. Class of Service Options

Enable the Public Network Access via DPNSS field in the class of service for all devices that make outgoing calls through SIP trunks, PRI trunks, LS trunks, and so forth that are connected to SIP Trunks.

9. (Optional) Set up Calling Party Number Substitution for outbound calls

- For DID-based substitution, add new rules in the DID Ranges for CPN Substitution form ( ), and then associate the rules to links in the SIP Peer Profile form ( ).

- For DN-based substitution, complete the Associated Directory Numbers form.

NOTE: You can also configure the CPN Substitution Number in the User and Services Configuration form.

10. (Optional) Set up Inward Dialing Modification for inbound calls:

11. Add new rules in the Inward Dialing Modification form .

12. Associate the rules to links in the SIP Peer Profile Called Party Inward Dialing Modification ( ) and SIP Peer Profile Calling Party Inward Dialing Modification ( ) forms

NOTE: You can associate Inward Dialing Modification rules to SIP devices in the SIP Device Capabilities form.

11. (Optional) To enable external hot desk users (EHDUs) to access mid-call features

Network Elements form

- Add an element (outbound proxy or session border controller) that supports KPML digit detection.

SIP Peer Profile form

- Program Authentication and Key Press Event options for the SIP trunk.

12. (Optional) To implement Call Billing for SIP Gateway

Network Zones form

- Configure a unique Default CPN and Default Billing Number for each zone in a cluster.

- Set the "Use P-Asserted-Identify for Billing" field to Yes (SIP Peer Profile form).

Class of Service Options form

- Enable option "Use default billable number for trunk billing" (

Associated Directory Numbers form

- If you want to assign a specific billing number and/or caller ID to a specific user, configure the (end-user switch). Alternatively, you can use the User and Services Configuration ( ) form to provision the user's CPN Substitution and Billing numbers.

SIP Peer Profile form

- If you wish to send both Default Billing Number and the user-specific billing number, set the "Use P-Preferred Identity Header" to "User Associated Billing".

For more information, see the Programming section in Call Billing for SIP Gateway.

Related Topics

- SIP Trunking for Service Provider

- SIP Call Processing Features

- Maintaining the SIP Interface

- SIP Satellite Office

- Call Billing for SIP Gateway