See the following sections in this chapter:
The Cisco Unified Communications Manager SIP trunk integration makes connections through a LAN or WAN. A gateway provides connections to the PSTN.
For a list of supported versions of Cisco Unified CM that are qualified to integrate with Cisco Unity Connection through a SIP trunk, see the SIP Trunk Compatibility Matrix: Cisco Unity Connection, Cisco Unified Communications Manager, and Cisco Unified Communications Manager Express at http://www.cisco.com/en/US/products/ps6509/products_device_support_tables_list.html.
This document applies only when Cisco Unity Connection is installed on a separate server from Cisco Unified CM. This document does not apply to the configuration in which Cisco Unity Connection is installed as Cisco Business Edition—on the same server with Cisco Unified CM.
The phone system sends the following information with forwarded calls:
- The extension of the called party
- The extension of the calling party (for internal calls) or the phone number of the calling party (if it is an external call and the system uses caller ID)
- The reason for the forward (the extension is busy, does not answer, or is set to forward all calls)
Cisco Unity Connection uses this information to answer the call appropriately. For example, a call forwarded to Cisco Unity Connection is answered with the personal greeting of the user. If the phone system routes the call without this information, Cisco Unity Connection answers with the opening greeting.
The Cisco Unified CM SIP trunk integration with Cisco Unity Connection provides the following features:
- Call forward to personal greeting
- Call forward to busy greeting
- Caller ID
- Easy message access (a user can retrieve messages without entering an ID; Cisco Unity Connection identifies a user based on the extension from which the call originated; a password may be required)
- Identified user messaging (Cisco Unity Connection automatically identifies a user who leaves a message during a forwarded internal call, based on the extension from which the call originated)
- Message waiting indication (MWI)
The functionality of this integration may be affected by the issues described below.
Use of Cisco Unified Survivable Remote Site Telephony (SRST) Router
When a Cisco Unified Survivable Remote Site Telephony (SRST) router is part of the network and the Cisco Unified SRST router takes over call processing functions from Cisco Unified CM (for example, because the WAN link is down), phones at a branch office can continue to function. In this situation, however, the integration features have the following limitations:
- Call forward to busy greeting —When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a call is forwarded from a branch office to Cisco Unity Connection, the busy greeting cannot play.
- Call forward to internal greeting —When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a call is forwarded from a branch office to Cisco Unity Connection, the internal greeting cannot play. Because the PSTN provides the calling number of the FXO line, the caller is not identified as a user.
- Call transfers —Because an access code is needed to reach the PSTN, call transfers from Cisco Unity Connection to a branch office will fail.
- Identified user messaging —When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a user at a branch office leaves a message or forwards a call, the user is not identified. The caller appears as an unidentified caller.
- Message waiting indication —MWIs are not updated on branch office phones, so MWIs will not correctly reflect when new messages arrive or when all messages have been listened to. We recommend resynchronizing MWIs after the WAN link is reestablished.
- Routing rules —When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a call arrives from a branch office to Cisco Unity Connection (either a direct or forwarded call), routing rules will fail.
When the Cisco Unified SRST router uses PRI/BRI connections, the caller ID for calls from a branch office to Cisco Unity Connection may be the full number (exchange plus extension) provided by the PSTN and therefore may not match the extension of the Cisco Unity Connection user. If this is the case, you can let Cisco Unity Connection recognize the caller ID by using alternate extensions.
Redirected Dialed Number Information Service (RDNIS) needs to be supported when using SRST.
For information on setting up Cisco Unified SRST routers, see the “Integrating Voice Mail with Cisco Unified SRST” chapter of the applicable Cisco Unified SRST System Administrator Guide at http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_installation_and_configuration_guides_list.html.
Impact of Non-Delivery of RDNIS on Voicemail Calls Routed via AAR
RDNIS needs to be supported when using Automated Alternate Routing (AAR).
AAR can route calls over the PSTN when the WAN is oversubscribed. However, when calls are rerouted over the PSTN, RDNIS can be affected. Incorrect RDNIS information can affect voicemail calls that are rerouted over the PSTN by AAR when Cisco Unity Connection is remote from its messaging clients. If the RDNIS information is not correct, the call will not reach the voicemail box of the dialed user but will instead receive the automated attendant prompt, and the caller might be asked to reenter the extension number of the party they wish to reach. This behavior is primarily an issue when the telephone carrier is unable to ensure RDNIS across the network. There are numerous reasons why the carrier might not be able to ensure that RDNIS is properly sent. Check with your carrier to determine whether it provides guaranteed RDNIS delivery end-to-end for your circuits. The alternative to using AAR for oversubscribed WANs is simply to let callers hear reorder tone in an oversubscribed condition.