When a call originates from MiVoice Business to the SIP service provider, it terminates on a SIP route. This SIP route is directed to the SIP Peer and the digits dialed are presented to the service provider.
When a call originates from the SIP service provider to MiVoice Business, it is treated as an external PSTN-type call. The calling line category of the incoming call is set to CLC_ISDN. This allows for external treatment of the call for features such as call forwarding, call rerouting, camp-on, recall, and so forth.
From Service Provider - the calling number and name are based on what is available in the SIP header: first from P-Asserted-Identity, second from P-Preferred Identity, and third from the From header line. In some cases, this is not the information of the calling party, but the Parent number from the remote PBX/ Softswitch placing the call.
To Service Provider - The calling party information may be configured on a per peer basis. By default, the caller's name and number are delivered but can be suppressed using the SIP Peer Profile form. Prior to programming calling party information, configure CPN Substitution. First, program a DID range with a corresponding CPN substitution number on the DID Ranges for CPN Substitution form. Then, select the index number for the DID range on the SIP Peer Profile form.
Privacy on Outgoing Calls - the existing privacy mechanisms for MiVoice Business can be used: feature access code for Name Suppression on Outgoing Trunk Call, feature access code for Private Caller (hides the caller's identity from the called party on a per call basis), and the Telephone Directory form using the Private field. The SIP Peer Profile form has an option for Calling Line ID Restriction. This option replaces the calling party name and number with anonymous for display.
Privacy on Incoming Calls - incoming calls can contain a privacy header that prevents the caller name and number from being presented to untrusted networks or devices.
Call Control supports the Calling Line ID Restriction (CLIR) notifications from SIP devices and trunks. When a SIP module detects the privacy indicator for a call originated from a SIP trunk or device, all data (including the privacy indicator) is passed to Call Control, which uses this information to provide correct and private data on the next leg of the call.
NOTE: If third-party elements are included in the network, they may not be able to process the privacy indicator and the private information may be exposed.
If you want to override delivery of Private/Hidden information, use the following options:
for devices: COS option "Suppress Delivery of Caller ID Display between sets - Override"
for SIP trunks: SIP Peer Profile option "Reject Incoming Anonymous Calls"
When a MiVoice Business user places a SIP Trunk on hold, MiVoice Business provides Music on Hold, if programmed. The feature behaves similarly to PRI calls.
NOTE: Hold-on-Hold whereby a user who has been placed on hold is able to select an another line and keep the first call connected is not supported. A SIP device user attempting to invoke Hold-on-Hold will lose (disconnect) the first call.
Call Transfer is allowed on SIP Trunks.
NOTE: The MiVoice Business system also supports Release Link or Two B-Channel Transfers using REFER messages.
SIP voice mail servers use call diversion capabilities. Call Diversion is allowed only to numbers defined in the Call Routing forms in the MiVoice Business System Administration Tool. If the SIP service provider sends a redirect request to MiVoice Business that cannot be routed, the call is cleared and the user receives a re-order tone.
Calls on SIP Trunks are re-directed when the Call Forwarding feature is enabled on the MiVoice Business destination. The feature behaves similarly to PRI calls.
The SIP Service Provider may use a TEL URI calling scheme to set up calls to MiVoice Business. The SIP Peer Profile for Incoming DID form contains the mapping of the dialed digits to the SIP Peer. This mapping allows the correct policy to be applied to the incoming call. Different policies can be assigned based on the digits dialed from the Service Provider.
The SIP Service Provider supports Direct Inward Dialing (DID) to MiVoice Business. Use the SIP Peer Profile for Incoming DID form to configure DID dialing. The incoming digits are limited to seven digits or less when translated by call control, and can terminate at stations, hunt groups, ACD paths, and speedcall. This feature operates the same as DID calling on PRI trunks. Trunk service options to insert or absorb digits work the same as they do on other trunk types.
SIP trunks support conferencing applications.
SIP trunks support EHDU users. SIP trunks use Keypad Markup Language (KPML) to relay digits for processing (mid-call feature). The relayed digits can be detected by a MiVoice Border Gateway (MBG) or a Session Border Controller (SBC) that supports KPML.
DTMF is used for feature interactions with some SIP services. The following DTMF modes are supported:
Regular RFC 4733 DTMF digits
SIP INFO messages containing DTMF digit information