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The documentation set for this product strives to use bias-free language. For the purposes of this documentation set, bias-free is defined as language that does not imply discrimination based on age, disability, gender, racial identity, ethnic identity, sexual orientation, socioeconomic status, and intersectionality. Exceptions may be present in the documentation due to language that is hardcoded in the user interfaces of the product software, language used based on RFP documentation, or language that is used by a referenced third-party product. Learn more about how Cisco is using Inclusive Language.
A phone system integration enables communication between Cisco Unity Connection and the phone system, providing users with the following features:
See the following sections for detailed information:
A phone system integration depends on the following components to be successful:
– Integration with Cisco Unified Communications Manager
– Digital Integration with Digital PIMG Units
– DTMF Integration with Analog PIMG Units
– Serial (SMDI, MCI, or MD-110) Integration with Analog PIMG Units
– TIMG Serial (SMDI, MCI, or MD-10) Integration
– PIMG/TIMG Integrations and Cisco Unified Communications Manager Through Cisco Unified SIP Proxy
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 11-1 shows the network connections used in an integration with Cisco Unified CM.
Figure 11-1 Connections for an Integration with Cisco Unified Communications Manager
See the “Integrating with Cisco Unified Communications Manager (Using SCCP or SIP)” section 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.
Figure 11-2 Connections for 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.
Figure 11-3 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 Cisco Unity Connection server through the LAN or WAN using Session Initialization Protocol (SIP). Figure 11-4 shows the connections for a serial integration using analog PIMG units.
Figure 11-4 Connections for a Serial (SMDI, MCI, or MD-110) 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 10.x, at http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/integration/guide/pimg/guide/cucintpimg/cucintpimg300.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 Cisco Unity Connection server through the LAN or WAN using Session Initialization Protocol (SIP). Figure 11-5 shows the connections for a serial integration using TIMG units.
Figure 11-5 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 11-6 shows the required connections for an in-band integration using TIMG units.
Figure 11-6 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 Cisco Unity Connection by acting as a SIP proxy. The Cisco Unified SIP Proxy uses a SIP trunk integration with Unity Connection. Figure 11-7 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.
Figure 11-7 Connections for PIMG/TIMG and Cisco Unified Communications Manager Through Cisco Unified SIP Proxy
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:
Cisco Unified Communications Manager SCCP and SIP trunk integrations can also provide the following call information:
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:
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:
Note If Extension A calls Extension B and if the Display Name or ASCII Alerting Name of Extension B contains the word “Voicemail”, then the call arrives on Cisco Unity Connection as a Direct Call rather than as a forwarded Call.
The following steps give an overview of a sample path that an external call can take when traveling from the phone system to a user.
1. For Cisco Unified Communications Manager, when an external call arrives, the gateway sends the call over the LAN or WAN to Cisco Unified CM. Cisco Unified CM routes the call to the Cisco Unity Connection voice mail pilot number.
For circuit-switched phone systems, when an external call arrives via the PSTN, TI/PRI, DID or LS/GS analog trunks, the phone system routes the call to the Cisco Unity Connection voice mail pilot number.
2. The phone system routes the call to an available Cisco Unity Connection voice messaging port.
3. Unity Connection answers the call and plays the opening greeting.
4. During the opening greeting, the caller enters an extension. For example, the caller enters 1234 to reach a person at that extension.
5. Unity Connection notifies the phone system that there is a call for extension 1234.
6. Depending on whether Unity Connection is set up to perform a release transfer or a supervised transfer, the following occurs:
For a detailed list of the requirements for a specific integration, see the applicable Cisco Unity Connection integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.
If Cisco Unity Connection is configured for a Unity Connection cluster, see the “Balancing the Load of Calls Unity Connection Servers Handle” section and the “Configuration for Dial-out Voice Messaging Ports” section.
In addition, consider the following list of integration issues:
Cisco Unity Connection and Cisco Unified Communications Manager deployment models, including single-site messaging, centralized messaging, and distributed messaging, can be combined to suit customer requirements. When choosing a deployment model, you must consider a range of issues, for example:
For a detailed explanation of deployment models and their relative merits, see the “Collaboration System Components and Architecture” chapter in Cisco Collaboration System 10.x Solution Reference Network Designs (SRND) available at http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/ovarchit.html.
Some deployment models, such as centralized messaging with distributed call processing, require placement of phones across the WAN from the Cisco Unity Connection server. When deploying phones across the WAN from the Unity Connection server, see the “Collaboration System Components and Architecture” chapter in Cisco Collaboration System 10.x Solution Reference Network Designs (SRND) available at http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/ovarchit.html.for guidance on capacity planning and call admission control (CAC) for these phones. When integrating Cisco Unity Connection with a circuit-switched phone system (TDM PBX), see the PIMG Integration Guide or the TIMG Integration Guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html for capacity planning for the PIMG/TIMG units deployed at these remote/branch sites to support phones at these sites.
Cisco Unity Connection supports Cisco Unified Communications Manager integrations through both SCCP and SIP interfaces. Table 11-1 shows the major differences in these integration methods.
Cisco Unified Communications Manager Session Manager Edition 8.5 is supported with Cisco Unity Connection 10.x. Cisco Unified Communications Manager-Session Manager Edition (Cisco UCM-SME) Release 8.5 uses SIP Early Offer with SIP trunk Normalization script to integrate with Third-party PBX's. For more information, see the interoperability portal for Third-party PBX's tested with Unity Connection via a SIP trunk using CUCM-SME and SIP normalization at
http://www.cisco.com/en/US/solutions/ns340/ns414/ns728/interOp_ucSessionMgr.html.
For information on the compatibility of Unity Connection and Cisco Unified CM versions, see the following documents:
For information on how to integrate Unity Connection with Cisco Unified CM, see the applicable Cisco Unity Connection integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.
For more information on using the SIP protocol to integrate Unity Connection with Cisco Unified CM, see the “Integrating Using SIP” section.
For information on the voice messaging solutions available for Cisco Unified CM, see the “Collaboration System Components and Architecture” chapter in Cisco Collaboration System 10.x Solution Reference Network Designs (SRND) available at http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/ovarchit.html.
A potential point of vulnerability for a Cisco Unity Connection system is the connection between Unity Connection and Cisco Unified Communications Manager. Possible threats include:
See the following sections for additional details:
Cisco Unified Communications Manager Release 4.1(3) or later for SCCP integrations or Cisco Unified Communications Manager Release 5.x or later for SIP trunk integrations can secure the connection with Cisco Unity Connection against security threats. The Cisco Unified CM security features that Unity Connection can take advantage of are described in Table 11-2 .
Note that Cisco Unified CM authentication and encryption protects only calls to Unity Connection. Messages that are recorded on Unity Connection are not protected by Cisco Unified CM authentication and encryption but can be protected by the Unity Connection secure messaging feature.
For more information on secure messaging, see the “Securing User Messages in Cisco Unity Connection 10.x” chapter of the Security Guide for Cisco Unity Connection Release 10.x, at http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/security/guide/10xcucsecx/10xcucsec070.html.
The security features (authentication and encryption) between Unity Connection and Cisco Unified CM require the following for SCCP integrations:
The process of authentication and encryption of Unity Connection voice messaging SCCP ports occurs as follows:
1. Each Unity Connection voice messaging port connects to the TFTP server, via TFTP port 69, downloads the CTL file, and extracts the certificates for all Cisco Unified CM servers.
2. Each Unity Connection voice messaging port establishes a network connection to the Cisco Unified CM TLS port. By default, the TLS port is 2443, though the port number is configurable.
3. Each Unity Connection voice messaging port establishes a TLS connection to the Cisco Unified CM server, at which time the device certificate is verified and the voice messaging port is authenticated.
4. Each Unity Connection voice messaging port registers with the Cisco Unified CM server, specifying whether the voice messaging port will also use media encryption.
The process of authentication and encryption of Unity Connection voice messaging SIP port groups occurs as follows:
1. Each Unity Connection voice messaging port group connects to the TFTP server, via TFTP port 69, downloads the CTL file, and extracts the certificates for all Cisco Unified CM servers.
2. Each Unity Connection voice messaging port group establishes a network connection to the Cisco Unified CM TLS port. By default, the TLS port is 2443, though the port number is configurable.
3. Each Unity Connection voice messaging port group establishes a TLS connection to the Cisco Unified CM server, at which time the device certificate is verified and the voice messaging port group is authenticated.
4. Each Unity Connection voice messaging port group registers with the Cisco Unified CM server, specifying whether the voice messaging port group will also use media encryption.
When a call is made between Cisco Unity Connection and Cisco Unified CM, the call-signaling messages and the media stream are handled in the following manner:
The Security Mode settings in Cisco Unity Connection Administration determine how the ports handle call-signaling messages and whether encryption of the media stream is possible. Table 11-3 describes the effect of the Security Mode settings on the Telephony Integrations > Port > Port Basics page for each port in an SCCP integration.
The authentication and encryption features between Cisco Unity Connection and Cisco Unified CM can be enabled and disabled by changing the Security Mode for all Cisco Unified CM clusters to Non-Secure, and by changing the applicable settings in the Cisco Unified Communications Manager Administration.
Authentication and encryption can be reenabled by changing the Security Mode to Authenticated or Encrypted.
Note After disabling or re-enabling authentication and encryption, it is not necessary to export the Unity Connection server root certificate and copy it to all Cisco Unified CM servers.
When Cisco Unity Connection has multiple Cisco Unified CM phone system integrations, each Cisco Unified CM phone system integration can have different Security Mode settings. For example, one Cisco Unified CM phone system integration can be set to Encrypted, and a second Cisco Unified CM phone system integration can be set to Non-Secure.
For troubleshooting purposes, authentication and encryption for Cisco Unity Connection voice messaging ports can be individually enabled and disabled. At all other times, the Security Mode setting for all individual voice messaging ports in a Cisco Unified CM port group should be the same.
The Real-Time Transport Protocol (RTP) is used to send and receive audio and video packets over the IP network. Each discrete packet has a fixed-size header, but the packets themselves can vary in size, depending on the size of the audio stream to be transported (which varies by codec) and the packetization setting. This variable size function helps utilize network bandwidth more efficiently. Reducing the number of packets that are created per call sends fewer total bytes over the network.
Packetization is set in the Cisco Unified CM Service Parameters, in the Preferred G711 Millisecond PacketSize and Preferred G729 Millisecond PacketSize parameters. Cisco Unity Connection supports any packet size up to 30ms for G.711 audio, and any packet size up to 60 ms for G.729a audio. The default setting is 20ms for both; there may be latency issues with lower settings.
DSCP is a priority setting on each packet. DSCP helps intermediary routers manage network congestion and lets them know which packets to prioritize ahead of others. Following Cisco AVVID standards, Unity Connection marks the SCCP and SIP packets (call control) with a default DSCP value of 24 (the TOS octet is 0x60), and the RTP packets (audio and video traffic) with a default DSCP value of 46 (the TOS octet is 0xB8). Thus, the RTP audio and video packets can be assigned priority over other packets using the router settings. Note that even though Cisco Unified CM allows you set different DSCP values, when integrated with Unity Connection, the DSCP values set by Unity Connection always take precedence. The marking of both SCCP and SIP packets is configurable in Unity Connection on the System Settings > Advanced > Telephony Configuration page in Cisco Unity Connection Administration.
With each new audio stream (once per call), Cisco Unified CM tells Unity Connection which packet size to use, and Unity Connection sets the DSCP priority for the stream. The entire stream (call) stays at the specified packet size and priority. For example, an audio stream could be broken up into packets of 30ms each. A 30ms G.729a audio stream would be 30 bytes plus the header per packet, and a 30ms G.711 stream would be 240 bytes plus the header per packet. For information on setting Cisco Unified CM Service Parameters, see the Cisco Unified CM documentation at http://www.cisco.com/en/US/products/sw/voicesw/ps556/tsd_products_support_series_home.html.
Note You can change the codecs that Unity Connection advertises on the Telephony Integrations > Port Group > Edit Codec Advertising configuration page in Cisco Unity Connection Administration.
For a discussion of supported advertised or “on the line” audio codecs and system-level recording audio codecs, see the “Audio Codecs” section.
For Cisco Unified Communications Manager SCCP integrations, when a Cisco Unified CM cluster is configured and Cisco Unified CM failover occurs as calls are in progress, the voice messaging ports may experience a delay registering with the secondary Cisco Unified CM server.
Unity Connection ports can register more quickly after Cisco Unified CM failover occurs if the port groups are configured as follows:
– The first port group contains half the voice messaging ports for the Cisco Unified CM integration (including answering and dialout ports) configured as described in the applicable chapter of the Cisco Unified Communications Manager SCCP Integration Guide for Cisco Unity Connection Release 10.x.
– The second port group contains the remaining half of the ports for the Cisco Unified CM integration (including answering and dialout ports) as described in the applicable chapter of the same guide.
Note The Cisco Unified Communications Manager SCCP Integration Guide for Cisco Unity Connection Release 10.x. is available at http://www.cisco.com/en/US/docs/voice_ip_comm/connection/10x/integration/guide/cucm_sccp/cucintcucmskinny.html.
– For the first port group, the Cisco Unified CM servers are listed in the order specified in the applicable chapter of the Cisco Unified Communications Manager SCCP Integration Guide for Cisco Unity Connection Release 10.x.
– For the second port group, the Cisco Unified CM servers are listed in the reverse order.
Cisco Unity Connection supports IPv4, IPv6, or dual-stack (IPv4 and IPv6) addressing with Cisco Unified Communications Manager phone system integrations via SCCP or SIP. When IPv6 is enabled, Connection can obtain an IPv6 address either through router advertisement, through DHCP, or by manually configuring an address either in Cisco Unified Operating System Administration or using the command-line interface.
For SCCP integrations with Cisco Unified CM, if Unity Connection is configured to listen for incoming IPv4 and IPv6 traffic, you can configure the addressing mode that Unity Connection uses for call control signaling for each port group to use either IPv4 or IPv6. (This mode is also used when connecting to a TFTP server.)
For SIP integrations with Cisco Unified CM, if Unity Connection is configured to listen for incoming IPv4 and IPv6 traffic, you can configure the addressing mode that Unity Connection uses for call control signaling for each port group to use either IPv4 or IPv6. (This mode is also used when connecting to a TFTP server.) In addition, you can configure the addressing mode that Unity Connection uses for media for each port group to use either IPv4 or IPv6.
IPv6 support is disabled by default. You can enable IPv6 and configure IPv6 address settings either in Cisco Unified Operating System Administration or in the CLI. For information on enabling and configuring IPv6 when setting up a new Cisco Unified CM integration, 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 the following considerations when deploying IPv6 for Cisco Unified CM integrations:
Cisco Unity Connection supports Cisco Unified Communications Manager Express integrations through both SCCP and SIP interfaces. Figure 11-8 shows the connections.
Figure 11-8 Cisco Unity Connection SCCP and SIP Connections to Cisco Unified Communications Manager Express Over a LAN
See Table 11-4 for information on the differences in these integration methods.
For information on the compatibility of Unity Connection and Cisco Unified Communications Manager Express versions, see the following:
For information on how to integrate Unity Connection with Cisco Unified CM Express, see the applicable Cisco Unity Connection integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.
For more information on using the SIP protocol to integrate Unity Connection with Cisco Unified CM Express, see the “Integrating Using SIP” section.
A single Cisco Unity Connection server can support multiple versions of Cisco Unified CM Express. The version of Unity Connection being used must support all versions of Cisco Unified CM Express. See the following documents:
A single, centralized Cisco Unity Connection server can be used by multiple Cisco Unified CM Express routers. This configuration requires that one Cisco Unified CM Express router be on the same LAN as the Unity Connection server, and that this Cisco Unified CM Express router register all Unity Connection voice messaging ports. This Cisco Unified CM Express router (the SIP MWI server) is a proxy server that relays SIP MWI messages between the Unity Connection server and all other Cisco Unified CM Express routers (the SIP MWI clients). Note that Unity Connection voice messaging ports register only with the SIP MWI server (the Cisco Unified CM Express router that is on the same LAN as the Unity Connection server), not with the SIP MWI clients. See Figure 11-9.
Figure 11-9 Connections between Multiple Cisco Unified CM Express Routers and a Single Cisco Unity Connection Server
For information on configuring Unity Connection to support multiple Cisco Unified CM Express routers, see the applicable Cisco Unity Connection integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.
A single Cisco Unity Connection server can support multiple versions of Cisco Unified Communications Manager and Cisco Unified Communications Manager Express. The Unity Connection version must support all versions of Cisco Unified CM and/or Cisco Unified CM Express. See the following documents:
Cisco Unified Survivable Remote Site Telephony (SRST) can direct calls to Unity Connection during Cisco Unified CM fallback. When the WAN is down and Unity Connection has Basic Rate Interface (BRI) or Primary Rate Interface (PRI) access to the Cisco Unified SRST system, Unity Connection uses ISDN signaling (see Figure 11-10).
Figure 11-10 Cisco Unified Communications Manager Fallback with BRI or PRI
When the WAN is down and Unity Connection has foreign exchange office (FXO) or foreign exchange station (FXS) access to a public switched telephone network (PSTN), Unity Connection uses in-band dual tone multifrequency (DTMF) signaling (see Figure 11-11).
Figure 11-11 Cisco Unified Communications Manager Fallback with PSTN
In both configurations, phone message buttons remain active and calls to busy or unanswered numbers are forwarded to Unity Connection. The installer must configure access from the dial peers to the voice-mail system, and establish routing to Unity Connection for busy and unanswered calls and for the message button.
If Unity Connection is accessed over FXO or FXS, you must configure instructions (DTMF patterns) for Unity Connection so it can access the correct voice-mail system mailbox.
When using Cisco Unified SRST with Unity Connection, the integration has the following limitations during a WAN outage:
When the Cisco Unified SRST router uses PRI or BRI connections, the caller ID for calls from a branch office to Unity Connection may be the full number (exchange plus extension) provided by the PSTN and therefore may not match the extension of the Unity Connection user. In this case, you can let Unity Connection recognize the caller ID using alternate extensions.
When using Cisco Unified SRST, Redirected Dialed Number Information Service (RDNIS) must be supported.
For information on setting up Cisco Unified SRST routers, see the “Integrating Voice Mail with Cisco Unified SRST” chapter of the Cisco Unified SRST System Administrator Guide at http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_installation_and_configuration_guides_list.html.
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.
Cisco Unity Connection Survivable Remote Site Voicemail (Unity Connection SRSV) is a backup voicemail solution that works in conjunction with Cisco Unified Survivable Remote Site Telephony (SRST) for providing voicemail service to a branch during WAN outages.
Unity Connection SRSV is used in the centralized Cisco Unified Communications Manager and Cisco Unity Connection environment with multiple branch offices or small sites. It provides limited voicemail and auto-attendant features that remain in synchronization with the central Unity Connection voicemail service so that when the WAN outage or failure occurs, the Unity Connection SRSV solution can provide voicemail service to the subscribers at the branch. However, as soon as the network is restored, all the voicemails received by the branch subscribers are automatically uploaded to the central Unity Connection voicemail server.
For more information on how to configure Cisco Unity Connection SRSV at central Connection location, see the Complete Reference Guide for Cisco Unity Connection Survivable Remote Site Voicemail (SRSV) for Release 10.x, available at http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/srsv/guide/10xcucsrsvx.html.
SIP (Session Initiation Protocol) is the Internet Engineering Task Force standard for multimedia calls over IP. SIP is a peer-to-peer, ASCII-based protocol that uses requests and responses to establish, maintain, and terminate calls (or sessions) between two or more end points. See Table 11-5 .
Cisco Unity Connection accepts calls from a proxy server. Unity Connection relies on a proxy server or call agent to authenticate calls.
SIP uses a request/response method to establish communications between various components in the network and to ultimately establish a conference (call or session) between two or more endpoints. A single call may involve several clients and servers.
Users in a SIP network are identified by:
When a user initiates a call, a SIP request typically goes to a SIP server (either a proxy server or a redirect server). The request includes the caller address (From) and the address of the called party (To).
SIP messages are in text format using ISO 10646 in UTF-8 encoding (like HTML). In addition to the address information, a SIP message contains a start-line specifying the method and the protocol, a number of header fields specifying call properties and service information, and an optional message body which can contain a session description.
Cisco Unity Connection supports the following SIP integrations:
Third-party SIP trunks are currently not supported.
For more information on configuring SIP trunks between Unity Connection and Cisco Unified CM or Cisco Unified CM Express, see the applicable SIP trunk integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.
Note Cisco Unity Connection extracts Caller Id from Remote Party Id field and FROM field in SIP Invite. In addition, when the Remote Party Id option is unchecked on CUCM SIP trunk and the FROM field is set to Anonymous in a SIP header, Connection treats the caller as unknown.
Cisco Unity Connection can integrate with circuit-switched phone systems using the PIMG or TIMG units (media gateways) between circuit-switched phone systems and IP networks.
For a list of circuit-switched phone systems supported with Unity Connection using PIMG and TIMG integrations, 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 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 “How a Phone System Integration Works” section.
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:
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 will connect to the four phone systems. For example:
Note that even though the PIMG units come with eight ports, fewer than eight ports can be used on each unit.
If PIMG units will be 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:
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:
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.
Cisco Unity Connection supports as many phone systems as needed up to the maximum number of ports supported per Unity Connection server or active/active server pair (a Unity Connection cluster). See the Multiple Phone System Integration Guide for Cisco Unity Connection Release 10.x at http://www.cisco.com/en/US/docs/voice_ip_comm/connection/10x/integration/guide/multiple_integration/cuc10xintmultiple.html.
Cisco Unity Connection has the following requirements for multiple phone system integrations:
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:
Unity Connection supports the following formats in the user portion of a URI (the portion before the @ symbol):
Unity Connection supports the following formats in the host portion of a URI (the portion after the @ symbol):
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.
Cisco Unity Connection supports centralized voice messaging through the phone system, which supports various inter-phone system networking protocols including proprietary protocols such as Avaya DCS, Nortel MCDN, or Siemens CorNet, and standards-based protocols such as QSIG or DPNSS. Note that centralized voice messaging is a function of the phone system and its inter-phone system networking, not voice mail. Unity Connection will support centralized voice messaging as long as the phone system and its inter-phone system networking are properly configured.
When discussing phone systems involved in centralized voice messaging, there are essentially two types:
Centralized voice messaging provides voice messaging services to all users in a networked phone system environment. Unity Connection can be hosted on a message center PINX and provide voice messaging services to all users in an enterprise assuming the message center PINX and all user PINX phone systems are properly networked.
For a centralized voice messaging configuration to exist, a suitable inter-phone system networking protocol must exist to deliver a minimum level of feature support, such as:
Other features may be required depending on how the voice messaging system is to be used. For example, if it is also serving as an automated attendant, path-replacement is needed as this feature prevents calls from hair-pinning.
Not all phone systems can serve as a message center PINX. In this case, customers may wish to consider relocating Unity Connection to Cisco Unified Communications Manager and have Cisco Unified CM act as the message center PINX with the circuit-switched phone system now acting as the user PINX.
For information on configuring Unity Connection in a centralized voice messaging environment to be hosted on Cisco Unified CM serving as the message center PINX, see the following:
Note that if customers are deploying centralized voice messaging with Unity Connection and a circuit-switched phone system, it is up to the customer to determine whether the circuit-switched phone system can serve as a message center PINX on which Unity Connection can be hosted. If so, the customer should also confirm that there is support for the desired features, for example, MWIs, transfer, divert, and path-replacement.
Inter-cluster trunks between Cisco Unified CM clusters can be QSIG-enabled using the Annex M.1 feature, which allows Unity Connection to integrate with a single Cisco Unified CM cluster. Ports in the cluster with which Unity Connection is integrated can be dedicated to turning MWIs on and off for phones in other clusters.
Cisco Unity Connection supports an integration with a QSIG-enabled phone system through a Cisco ISR voice gateway. See Figure 11-12.
Figure 11-12 Connections Between the Phone System and Cisco Unity Connection
For more information on integrating Unity Connection with a QSIG-enabled phone system using Cisco ISR voice gateways, see the QSIG-Enabled Phone System with Cisco ISR Voice Gateway Integration Guide for Cisco Unity Connection 10.x at http://www.cisco.com/en/US/docs/voice_ip_comm/connection/10x/integration/guide/sip-qsig_gw/cuc10xintqsig.html.
For a list of all supported versions of Cisco Unified Communications Manager and Cisco Unified CM Express, see the applicable document, depending on the integration type:
For the most current list of other supported phone system integrations, see the applicable Cisco Unity Connection integration guide at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.
Unity Connection can integrate with one or more phone systems at the same time. For details, see the Multiple Phone System Integration Guide for Cisco Unity Connection Release 10.x at http://www.cisco.com/en/US/docs/voice_ip_comm/connection/10x/integration/guide/multiple_integration/cuc10xintmultiple.html.