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148 Authorized Self-Study Guide: Cisco Voice over IP (CVOICE) Note The T control character indicates that the destination-pattern value is a variablelength dial string. Using this control character enables the router to wait until all digits are received before routing the call. Dial-peer configuration is covered in the section, “Introducing Dial Peers.” Step 8. Specify the voice port associated with this dial peer. Router(config-dialpeer)#port 0/0/0 Example 3-2 shows the complete FXO voice port configuration. Example 3-2 FXO Voice Port Configuration Router(config)#voice-port 0/0/0 Router(config-voiceport)#signal groundstart Router(config-voiceport)#connection plar opx 4001 Router(config)#dial-peer voice 90 pots Router(config-dialpeer)#destination-pattern 9T Router(config-dialpeer)#port 0/0/0 E&M Voice Port Configuration Configuring an E&M analog trunk is straightforward. Three key options have to be set: ■ The signaling E&M signaling type ■ Two- or four-wire operation ■ The E&M type As an example, consider the topology shown in Figure 3-17. 2001 E&M Trunk Wink Start Type I Two-Wire 1001 2002 PBX 1002 Inbound DNIS Outbound DNIS E&M 1/1/1 2003 1003 2004 Figure 3-17 E&M Configuration Topology Chapter 3: Routing Calls over Analog Voice Ports In this example, you have been assigned to configure a voice gateway to work with an existing PBX system according to network requirements. You must set up a voice gateway to interface with a PBX to allow the IP phones to call the POTS phones using a four-digit extension. The configuration requirements are the following: ■ Configure the voice port to use wink-start signaling. ■ Configure the voice port to use 2-wire operation mode. ■ Configure the voice port to use Type I E&M signaling. ■ Configure a standard dial peer for the POTS phones behind the PBX. Both sides of the trunk need to have a matching configuration. The following example configuration shows an E&M trunk using wink-start signaling, E&M Type I, and twowire operation. Because E&M supports inbound and outbound DNIS, DID support is also configured on the corresponding dial peer. You could then complete the following steps to configure the E&M voice port: Step 1. Enter voice-port configuration mode. Step 2. Select the access signaling type to match the telephony connection you are making. Router(config-voiceport)#signal wink-start Step 3. Select a specific cabling scheme for the E&M port. Router(config-voiceport)#operation 2-wire Note This command affects only voice traffic. If the wrong cable scheme is specified, the user might get voice traffic in only one direction. Also, using this command on a voice port changes the operation of both voice ports on a voice port module (VPM) card. The voice port must be shut down and then opened again for the new value to take effect. Step 4. Specify the type of E&M interface. Router(config-voiceport)#type 1 Step 5. Activate the voice port. Router(config-voiceport)#no shutdown Step 6. Exit voice port configuration mode. Router(config-voiceport)#exit 149 150 Authorized Self-Study Guide: Cisco Voice over IP (CVOICE) Step 7. Create a dial peer for the POTS phones. Router(config)#dial-peer voice 10 pots Step 8. Specify the destination pattern for the POTS phones. Router(config-dialpeer)#destination-pattern 1... Step 9. Specify direct inward dial. Router(config-dialpeer)#direct-inward-dial Note DID is needed when POTS phones call IP Phones. In this case we match the POTS dial peer. This same dial peer is also used to call out to POTS phones. Step 10. Specify digit forwarding all, so that no digits will be stripped as they are forwarded out of the voice port. By default, only digits matched by wildcard characters in the destination-pattern command are forwarded. Router(config-dialpeer)#forward-digits all Step 11. Specify the voice port associated with this dial peer. Router(config-dialpeer)#port 1/1/1 Example 3-3 shows the complete E&M voice port configuration. Example 3-3 E&M Voice Port Configuration Router(config)#voice-port 1/1/1 Router(config-voiceport)#signal wink-start Router(config-voiceport)#operation 2-wire Router(config-voiceport)#type 1 Router(config-voiceport)#no shutdown Router(config-voiceport)#exit Router(config)#dial-peer voice 10 pots Router(config-dialpeer)#destination-pattern 1... Router(config-dialpeer)#direct-inward-dial Router(config-dialpeer)#forward-digits all Router(config-dialpeer)#port 1/1/1 Trunks Trunks are used to interconnect gateways or PBX systems to other gateways, PBX systems, or the PSTN. A trunk is a single physical or logical interface that contains several physical interfaces and connects to a single destination. This could be a single FXO port Chapter 3: Routing Calls over Analog Voice Ports that provides a single line connection between a Cisco gateway and a FXS port of small PBX system, a POTS device, or several T1 interfaces with 24 lines each in a Cisco gateway providing PSTN lines to several hundred subscribers. Trunk ports can be analog or digital and use a variety of signaling protocols. Signaling can be done using either the voice channel (in-band) or an extra dedicated channel (outof-band). The available features depend on the signaling protocol in use between the devices. Figure 3-18 illustrates a variety of possible trunk connections. T1 PRI Chicago E&M Trunk V PSTN San Jose T1 QSIG Trunk T1 CAS Trunk E1 R2 Trunk E1 CCS Trunk V Denver T1 QSIG Trunk London Rome V V T1 PRI Figure 3-18 E&M Trunks Consider the following characteristics of the trunks depicted in Figure 3-18: ■ If a subscriber at the London site places a call to the PSTN, the gateway uses one voice channel of the E1 R2 trunk interface. ■ If a subscriber of the legacy PBX system at the Chicago site needs to place a call to a subscriber with an IP phone connected to the Chicago gateway, the call will go via the E&M trunk between the legacy PBX and the gateway. ■ The Denver and the Chicago sites are connected to San Jose via Q Signaling (QSIG) to build up a common private numbering plan between those sites. Because Denver’s Cisco IP telephony rollout has not started yet, the QSIG trunk is established directly between San Jose’s gateway and Denver’s legacy PBX. 151 152 Authorized Self-Study Guide: Cisco Voice over IP (CVOICE) Analog Trunks Because many organizations continue to use analog devices, a requirement to integrate analog circuits with VoIP or IP telephony networks still exists. To implement a Cisco voice gateway into an analog trunk environment, the FXS, FXO, DID, and E&M interfaces are commonly used, as illustrated in Figure 3-19. Station Port FXS FXO Port Port FXS Port V FXS Port V PSTN FXO Port CO FXS Interface FXO Interface Trunk Side of PBX DID Port V PSTN V E&M Port CO E&M Interface Figure 3-19 DID Interface Analog Trunks PSTN carriers typically offer analog trunk features that can be supported on home phones. Table 3-5 presents a description of the common analog trunk features. Table 3-5 Analog Trunk Features Feature Description Caller ID Caller ID allows users to see the calling number before answering the phone. Message waiting Two methods activate an analog message indicator: ■ High-DC voltage message-waiting indicator (MWI) light and frequency-shift keying (FSK) messaging. ■ Stuttered dial tone for phones without a visual indicator. Call waiting When a user is on a call and a new call comes in, the user hears an audible tone and can “click over” to the new caller. Caller ID on call waiting When a user is on a call, the name of the second caller is announced or the caller ID is shown. Chapter 3: Routing Calls over Analog Voice Ports Table 3-5 Analog Trunk Features (continued) Feature Description Transfer This feature includes both blind and supervised transfers using the standard established by Bellcore laboratories. The flash hook method is common with analog trunks. Conference Conference calls are initiated from an analog phone using flash hook or feature access codes. Speed dial A user can set up keys for commonly dialed numbers and dial these numbers directly from an analog phone. Call forward all Calls can be forwarded to a number within the dial plan. Redial A simple last-number redial can be activated from analog phones. DID Supported on E&M and FXS DID ports. Figure 3-20 shows small business voice networks connected through a gateway to the PSTN. The voice network supports both analog phones and IP phones. The connection to the PSTN is through an FXO port, and the analog phone is connected to the small business network through an FXS port. The issue in this scenario is how the caller ID is passed to call destinations. Ext. 0113 Caller ID Display Number 555-0112 Name John Smith Caller ID Display Number 408 555-0100 Name ACME Enterprises 408 555-9999 Call 1 PSTN V Service Provider Database Number 408 555-0100 Name ACME Enterprises Figure 3-20 Analog Trunks - Example Call 2 Analog Extension Station ID Number 555-0112 Station ID Name John Smith 153 154 Authorized Self-Study Guide: Cisco Voice over IP (CVOICE) This example describes two calls; the first call is to an on-premises destination, and the second call is to an off-premises destination: ■ Call 1: Call 1 is from the analog phone to another phone on the premises. The FXS port is configured with a station ID name and station ID number. The name is John Smith, and the number is 555-0212. When a call is placed from the analog phone to another phone on the premises, an IP phone in this case, the caller name and number are displayed on the screen of the IP phone. ■ Call 2: Call 2 is placed from the same analog phone, but the destination is off the premises on the PSTN. The FXO port forwards the station-ID name and station-ID number to the CO switch. The CO switch discards the station ID name and station ID number and replaces them with information it has configured for this connection. For inbound calls, the caller ID feature is supported on the FXO port in the gateway. If the gateway is configured for H.323, the caller ID is displayed on the IP phones and on the analog phones (if supported). Note Although the gateway supports the caller ID feature, Cisco Unified Communications Manager does not support this feature on FXO ports if the gateway is configured for Media Gateway Control Protocol (MGCP). Centralized Automated Message Accounting A Centralized Automated Message Accounting (CAMA) trunk is a special analog trunk type originally developed for long-distance billing but now mainly used for emergency call services (911 and E911 services). You can use CAMA ports to connect to a Public Safety Answering Point (PSAP) for emergency calls. A CAMA trunk can send only outbound automatic number identification (ANI) information, which is required by the local public safety answering point (PSAP). CAMA interface cards and software configurations are targeted at corporate enterprise networks and at service providers and carriers who are creating new or supplementing existing networks with Enhanced 911 (E911) services. CAMA carries both calling and called numbers by using in-band signaling. This method of carrying identifying information enables the telephone system to send a station identification number to the PSAP via multifrequency (MF) signaling through the telephone company E911 equipment. CAMA trunks are currently used in 80 percent of E911 networks. The calling number is needed at the PSAP for two reasons: ■ The calling number is used to reference the Automatic Location Identification (ALI) database to find the exact location of the caller and any extra information about the caller that might have been stored in the database. Chapter 3: Routing Calls over Analog Voice Ports ■ Note The calling number is used as a callback number in case the call is disconnected. A number of U.S. states have initiated legislation that requires enterprises to connect directly to the E911 network. The U.S. Federal Communications Commission (FCC) has announced model legislation that extends this requirement to all U.S. states. Enterprises in areas where the PSTN accepts 911 calls on ISDN trunks can use existing Cisco ISDN voice-gateway products because the calling number is an inherent part of ISDN. You must check local legal requirements when using CAMA. Calls to emergency services are routed based on the calling number, not the called number. The calling number is checked against a database of emergency service providers that cross-references the service providers for the caller location. When this information is determined, the call is then routed to the proper PSAP, which dispatches services to the caller location. During the setup of an E911 call, before the audio channel is connected, the calling number is transmitted to each switching point, known as a selective router, via CAMA. The VIC2-2FXO and VIC2-4FXO cards support CAMA via software configuration. CAMA support is also available for the Cisco 2800 Series and 3800 Series ISRs. It is common for E911 service providers to require CAMA interfaces to their network. Figure 3-21 shows a site that has a T1 PRI circuit for normal inbound and outbound PSTN calls. Because the local PSAP requires a dedicated CAMA trunk for emergency (911) calls, all emergency calls are routed using a dial peer pointing to the CAMA trunk. Austin T1 PRI for Standard Calls 0/0/0 PSTN 1/1/1 CAMA Trunk for Emergency Calls Figure 3-21 Configuring a CAMA Trunk PSAP 155 156 Authorized Self-Study Guide: Cisco Voice over IP (CVOICE) The voice port 1/1/1 is the CAMA trunk. The actual configuration depends on the PSAP requirements. In this case, the digit 1 is used to signal the area code 312. The voice port is then configured for CAMA signaling using the signal cama command. Five options exist: ■ KP-0-NXX-XXXX-ST: 7-digit ANI transmission. The Numbering Plan Area (NPA), or area code, is implied by the trunk group and is not transmitted. ■ KP-0-NPA-NXX-XXXX-ST: 10-digit transmission. The E.164 number is fully transmitted. ■ KP-0-NPA-NXX-XXXX-ST-KP-YYY-YYY-YYYY-ST: Supports CAMA signaling with ANI/Pseudo ANI (PANI). ■ KP-2-ST: Default transmission when the CAMA trunk cannot get a corresponding Numbering Plan Digit (NPD) in the look-up table or when the calling number is fewer than 10 digits. (NPA digits are not available.) ■ KP-NPD-NXX-XXXX-ST: 8-digit ANI transmission, where the NPD is a single MF digit that is expanded into the NPA. The NPD table is preprogrammed in the sending and receiving equipment (on each end of the MF trunk). For example: 0=415, 1=510, 2=650, 3=916 05551234 = (415) 555-1234, 15551234 = (510) 555-1234 The NPD value range is 0–3. When you use the NPD format, the area code needs to be associated with a single digit. You can preprogram the NPA into a single MF digit using the ani mapping voice port command. The number of NPDs programmed is determined by local policy as well as by the number of NPAs the PSAP serves. Repeat this command until all NPDs are configured or until the NPD maximum range is reached. In this example, the PSAP expects NPD signaling, with the area code 312 being represented by the digit 1. You could then complete the following steps to configure the voice port for CAMA operation: Step 1. Configure a voice port for 911 calls. Router(config)#voice-port 1/1/1 Router(config-voiceport)#ani mapping 1 312 Router(config-voiceport)#signal cama kp-npd-nxx-xxxx-st Chapter 3: Routing Calls over Analog Voice Ports Step 2. Configure a dedicated dial peer to route emergency calls using the CAMA trunk when a user dials “911.” Router(config)#dial-peer voice 911 pots Router(config-dialpeer)#destination-pattern 911 Router(config-dialpeer)#prefix 911 Router(config-dialpeer)#port 1/1/1 Step 3. Configure a dedicated “9911” dial peer to route all emergency calls using the CAMA trunk when a user dials “9911.” Router(config)#dial-peer voice 9911 pots Router(config-dialpeer)#destination-pattern 9911 Router(config-dialpeer)#prefix 911 Router(config-dialpeer)#port 1/1/1 Step 4. Configure a standard PSTN dial peer for all other inbound and outbound PSTN calls. Router(config)#dial-peer voice 910 pots Router(config-dialpeer)#destination-pattern 9[2-8]......... Router(config-dialpeer)#port 0/0/0:23 Example 3-4 shows the complete CAMA trunk configuration. Example 3-4 CAMA Trunk Configuration Router(config)#voice-port 1/1/1 Router(config-voiceport)#ani mapping 1 312 Router(config-voiceport)#signal cama KP-NPD-NXX-XXXX-ST Router(config)#dial-peer voice 911 pots Router(config-dialpeer)#destination-pattern 911 Router(config-dialpeer)#prefix 911 Router(config-dialpeer)#port 1/1/1 Router(config)#dial-peer voice 9911 pots Router(config-dialpeer)#destination-pattern 9911 Router(config-dialpeer)#prefix 911 Router(config-dialpeer)#port 1/1/1 Router(config)#dial-peer voice 910 pots Router(config-dialpeer)#destination-pattern 9[2-8]......... Router(config-dialpeer)#port 0/0/0:23 Direct Inward Dial Typically, FXS ports connect to analog phones, but some carriers offer FXS trunks that support DID. The DID service is offered by telephone companies, and it enables callers to dial an extension directly on a PBX or a VoIP system (for example, Cisco Unified 157