Cisco Unified Communications Manager, Cisco Unified Communications Manager Express, and SIP proxy servers use network connections that carry all communication to and from Cisco Unity Connection. Figure shows the network connections used in an integration with Cisco Unified CM.

See the Integrating with Cisco Unified Communications Manager Using SCCP or SIP for additional information.

The phone system sends call information, MWI requests, and voice connections through the digital lines, which connect the phone system to the PIMG units (media gateways). The PIMG units communicate with the Cisco Unity Connection server through the LAN or WAN using Session Initiation Protocol (SIP). Figure 11-2 shows the connections used in a digital integration using digital PIMG units.

The phone system sends call information, MWI requests, and voice connections through the analog lines, which connect the phone system to the PIMG units (media gateways). The PIMG units communicate with the Cisco Unity Connection server through the LAN or WAN using Session Initiation Protocol (SIP). Figure 11-3 shows the connections for a DTMF integration using analog PIMG units.

The phone system sends call information and MWI requests through the data link, which is an RS-232 serial cable that connects the phone system and the master PIMG unit (media gateways). Voice connections are sent through the analog lines between the phone system and the PIMG units. The PIMG units communicate with the Unity Connection server through the LAN or WAN using Session Initialization Protocol (SIP). Figure shows the connections for a serial integration using analog PIMG units.

Note

When you use multiple PIMG units, one PIMG unit must be designated the master PIMG unit, which is connected to the serial cable from the phone system. It is not possible to “daisy chain” the serial ports on the PIMG units. You can add a secondary master PIMG unit to an integration. For details, see the “Adding a Secondary Master PIMG Unit” chapter of the PIMG Integration Guide for Cisco Unity Connection Release 12.x, at https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/12x/integration/guide/pimg/b_12xcucintpimg.html.

The TIMG integration uses one or more TIMG units between circuit-switched phone systems and IP networks. On the circuit-switched phone system side, there is a T1-CAS interface. On the IP side, there is a SIP interface, which is how Cisco Unity Connection communicates with the TIMG units. To Unity Connection, the integration is essentially a SIP integration. Unity Connection communicates with the TIMG units over the IP network using SIP and RTP protocols. The TIMG units communicate with the circuit-switched phone system over the phone network using serial protocols (SMDI, MCI, or MD-110).

The phone system sends call information and MWI requests through the data link, which is an RS-232 serial cable that connects the phone system and the master TIMG unit. Voice connections are sent through the T1 digital lines between the phone system and the TIMG units. The TIMG units communicate with the Unity Connection server through the LAN or WAN using Session Initialization Protocol (SIP). Figure shows the connections for a serial integration using TIMG units.

The phone system sends call information, MWI requests, and voice connections through the T1 digital lines, which connect the phone system and the TIMG units. The TIMG units communicate with the Cisco Unity Connection server through the LAN or WAN using Session Initialization Protocol (SIP). Figure shows the required connections for an in-band integration using TIMG units.

The Cisco Unified SIP Proxy allows the PIMG/TIMG integrations and Cisco Unified Communications Manager to share the same voice messaging ports on Unity Connection by acting as a SIP proxy. The Cisco Unified SIP Proxy uses a SIP trunk integration with Unity Connection. Figure shows the connections. For more information, see the application notes for Cisco Unified SIP Proxy at http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/interOp_sipProxy.html.

For an integration to be successful, Unity Connection and the phone system must know the connections to use (for example, IP addresses and channels) and the expected method of communication (for example, IP packets, serial packets, and DTMF tones). Certain integrations require specific codes or extensions for turning MWIs on and off.

There are required settings for Unity Connection, and programming for the phone system, that must be made in order to enable the integration. For information on these settings, see the applicable Cisco Unity Connection integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.

The phone system and Unity Connection exchange call information to manage calls and to make the integration features possible. With each call, the following call information is typically passed between the phone system and Unity Connection:

• 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 phone system supports caller ID).

• The reason for the forward (the extension is busy, does not answer, or is set to forward all calls). There is also a reason code for Direct Calls.

Cisco Unified Communications Manager SCCP and SIP trunk integrations can also provide the following call information:

• Called number

• First redirecting number

• Last redirecting number

  Note

  Unity Connection can use either the first redirecting number or last redirecting number, depending on the setting of the Use Last (Rather than First) Redirecting Number for Routing Incoming Call check box on the System Settings > Advanced > Conversations page in Cisco Unity Connection Administration.

If the phone system sends the necessary information and if Unity Connection is configured correctly, an integration can provide the following integration functionality:

• Call forward to personal greeting

• Call forward to busy greeting

• Caller ID

• Easy message access (a user can retrieve messages without entering an ID because Unity Connection identifies the user based on the extension from which the call originated; a password may be required)

• Identified user messaging (Unity Connection identifies the user who leaves a message during a forwarded internal call, based on the extension from which the call originated)

The phone system uses a set of signals to set up, monitor, and release connections for a call. Cisco Unity Connection monitors call control signals to determine the state of the call, and uses these signals to respond appropriately to phone system actions and to communicate with the phone system. For example, a caller who is recording a message might hang up, so Unity Connection detects that the call has ended and stops recording.

Depending on the phone system, the following types of call control signals are used:

RDNIS must 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 voice mail 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 caller does not reach the mailbox of the dialed user but instead hears the automated attendant prompt, and might be asked to reenter the extension number of the party the caller wants to reach. This behavior is primarily an issue when the phone 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.

Cisco Unity Connection supports a topology with centralized call processing and distributed messaging, in which your Unity Connection server is located at a remote site or branch office and registered with Cisco Unified CM at a central site.

When the WAN link fails, the phones fall back to the Cisco Unified CM Express-as-SRST device. Unity Connection can also fall back to the Cisco Unified CM Express-as-SRST device, which lets users at the remote site access their voice messages and see message waiting indicators (MWIs) during a WAN outage. Note that MWIs must be resynchronized from the Unity Connection server whenever a failover happens from Cisco Unified CM to Cisco Unified CM Express-as-SRST or vice versa.

For information on setting up this configuration, see the Integrating Cisco Unity Connection with Cisco Unified CME-as-SRST configuration guide at http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_installation_and_configuration_guides_list.html.

The PIMG integration uses one or more PIMG units between the circuit-switched phone systems and IP network. On the circuit-switched phone system side, there are both digital (feature-set) and analog interfaces; the interface used depends on the phone system to which Cisco Unity Connection is connected. On the IP side, there is a SIP interface, which is how Unity Connection communicates with the PIMG units. To Unity Connection, the integration is essentially a SIP integration. Unity Connection communicates with the PIMG units over the IP network using SIP and RTP protocols. The PIMG units communicate with the circuit-switched phone system over the phone network using phone system-specific protocols (digital, analog, or serial).

For high-level descriptions of each PIMG integration type, and illustrations showing the network connections, see the Working of a Phone System Integration.

For PIMG/TIMG setup and configuration, the installer does the following steps as documented in the applicable integration guide:

1. Configure the phone system.
2. Configure the PIMG/TIMG units. PIMG/TIMG settings are somewhat phone system-specific, but less so than phone system configuration.
3. Configure Cisco Unity Connection for the integration.

For information on configuring the phone system, PIMG/TIMG units, and Unity Connection, see the applicable Cisco Unity Connection integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.

Note that when receiving shipment of PIMG or TIMG units, it may be necessary to update the firmware on the units. The PIMG/TIMG Administration interface provides a simple method to update the firmware files. Firmware updates are available at http://tools.cisco.com/support/downloads/go/Redirect.x?mdfid=278875240 (note that you must sign in to www.cisco.com to access the URL). For details, see the applicable integration guide.

Cisco Unity Connection supports the following serial protocols:

• SMDI

• MCI

• MD-110

The serial port on PIMG/TIMG units was originally designed as a management port rather than as a standard RS-232 serial port. Consequently, a custom serial cable (which is available from Cisco) is necessary for the data link between the phone system and the master PIMG/TIMG unit.

PIMG units have eight ports. To increase system port capacity, multiple PIMG units can be stacked. For example, if 32 ports are needed, four PIMG units can be stacked.

TIMG units, which integrate with circuit-switched phone systems that support T1-CAS, have 24 T1 ports per span in a single rack-optimized unit. Single-span, dual-span, and quad-span TIMG units are available.

PIMG/TIMG integrations support Unity Connection clusters (active/active high availability). Configuration changes are required both for the PIMG/TIMG units and for the Unity Connection servers, as described in the applicable Cisco Unity Connection integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.

PIMG/TIMG units can be separated by a WAN to support circuit-switched phone systems at remote branch office sites. For example, Cisco Unity Connection could be placed at a centralized headquarters and support circuit-switched phone systems both at the headquarters and at the branch office sites.

As an example, assuming there are four phone systems from four different manufacturers (for example, Nortel, Avaya, NEC, and Siemens), four different phone system integrations could be created on the Unity Connection server to support the four phone systems. A standalone Unity Connection server supports up to 144 ports that connects to the four phone systems. For example:

• At the Seattle site, 15 PIMG units can be stacked to support 120 ports.

• At the New York site, two PIMG units can be stacked to support 16 ports.

• At the Tokyo site, one PIMG unit can be used to support four ports.

• At the Dallas site, one PIMG unit can be used to support two ports.

Note that even though the PIMG units come with eight ports, fewer than eight ports can be used on each unit.

If PIMG units are separated by a WAN to support remote phone systems, correct audio codec selection, bandwidth capacity planning, and QOS planning are required. Both the G.729a and G.711 audio codecs are supported by PIMG units and by Unity Connection. Because PIMG units are Dialogic devices rather than Cisco devices, the use of location-based CAC is not applicable. The following network and bandwidth requirements are required when placing the PIMG across a WAN:

• For the G.729a audio codec, a minimum of 32.76 Kbps (assumes Ethernet, payload of 20 bytes, 5 percent overhead) guaranteed bandwidth for each voice messaging port.

• For the G.711 audio codec, a minimum of 91.56 Kbps (assumes Ethernet, payload of 160 bytes, 5 percent overhead) guaranteed bandwidth for each voice messaging port.

• No network devices that implement network address translation (NAT).

When PIMG units are separated by a WAN, prioritize your call control and media traffic through proper QOS traffic, marking for voice traffic originating on the PIMG units. Set the Call Control QOS Byte and RTP QOS Byte on PIMG units to the following values:

• In the Call Control QOS Byte field, enter 104.

• In the RTP QOS Byte field, enter 184.

Note that the Call Control QOS Byte and RTP QOS Byte fields on PIMG units define a decimal value that represents QOS bit flags. These values can be interpreted as either IPv4 TOS or Differentiated Services Codepoint (DSCP). For more details, see the Dialogic 1000 and 2000 Media Gateway Series User’s Guide, provided by Dialogic.

Unity Connection has the following requirements for multiple phone system integrations:

• All phone system and Unity Connection server requirements have been met. See the applicable Cisco Unity Connection integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.

• There must be an adequate number of voice messaging ports on the Unity Connection server to connect to the phone systems.

In addition to the primary extension for each user, you can set up alternate extensions. Alternate extensions can be used for various reasons, such as handling multiple line appearances on user phones. Alternate extensions can also make calling Cisco Unity Connection from an alternate device—such as a mobile phone, a home phone, or a phone at another work site—more convenient.

When you specify the phone number for an alternative extension, Unity Connection handles all calls from that number in the same way that it handles calls from a primary extension (assuming that ANI or caller ID is passed along to Unity Connection from the phone system). This means that Unity Connection associates the alternate phone number with the user account, and when a call comes from that number, Unity Connection prompts the user to enter a password and sign in.

Unity Connection supports dialing using URIs for alternate extensions. URIs look like email addresses and follow the username@host format where the host portion is an IPv4 address or a fully qualified domain name. A URI is a uniform resource identifier, a string of characters that can be used to identify a directory number. If that directory number is assigned to a phone then Cisco Unity Connection can route calls to that phone using the URI. URI dialing is available for both SIP endpoints that support URIs.

The administrator can import the end users directory URI into Unity Connection from the LDAP directory or Cisco Unified Communications Manager.

Note	In HTTPS, CCI, and Diginet networking, URI for alternate extensions is replicated only on the nodes that support directory URI.

URIs are alphanumeric strings consisting of a user and a host address separated by the @ symbol. The URI field has a maximum length of 40 characters.

Unity Connection supports the following formats for URIs:

• user@domain (for example, joe@cisco.com)

• user@ip_address (for example, joe@10.10.10.1)

Unity Connection supports the following formats in the user portion of a URI (the portion before the @ symbol):

• Accepted characters are a-z, A-Z, 0-9, !, $, %, &, *, _, +, ~, -, =, \, ?, \, ', ,, ., /, "", {}, []. <>.

• The user portion is case sensitive.

Unity Connection supports the following formats in the host portion of a URI (the portion after the @ symbol):

• Supports IPv4 addresses or fully qualified domain names.

• Accepted characters are a-z, A-Z, 0-9, hyphens, and dots.

• The host portion cannot start or end with a hyphen.

• The host portion cannot have two dots in a row.

• Minimum of one character.

• The host portion is case sensitive.

  Note	Use lower case for URIs.

You can set up Cisco Unity Connection to activate alternate MWIs when you want a new message for a user to activate the MWIs at up to 10 extensions. For example, a message left at extension 1001 can activate the MWIs on extensions 1001 and 1002.

Unity Connection uses MWIs to alert the user to new voice messages. MWIs are not used to indicate new email, fax, or return receipt messages.