Cisco IOS PSTN Telephony Interfaces

You can add numerous modular cards to your Cisco CME router to support PSTN connections of the types discussed in the preceding section. These technologies and hardware cards are not particular to Cisco CME. They can be used on any Cisco router platform that supports the card in question, independent of whether Cisco CME is enabled on the router. For example, you may choose to have two separate routers in your officeone configured for Cisco CME and the other as the PSTN voice gateway. Or you can combine both functions in the same router.

The following sections cover these hardware choices in greater detail, first looking at analog and then at digital PSTN trunks.

Analog Trunks

Voice interfaces range from two- and four-port FXO/FXS/E&M/DID cards up to 96/120-channel quad T1/E1 interfaces. The physical telephony interface for analog and BRI ports is provided by a plug-in voice interface card (VIC) and for a T1/E1 port by a voice or WAN interface card (VWIC).

Using various combinations of VICs and VWICs on a Cisco IOS router, you can build a Cisco CME system that includes a range of physical telephone interfaces. You can assemble a small analog telephony system with a few FXO ports used to connect to PSTN subscriber lines, or you can use digital telephony interfaces such as T1/E1 and ISDN BRI/PRI, or any combination of these. The specific hardware cards offering analog trunk and station (analog phone or fax machine) interfaces are discussed next.

Analog Trunk and Station Hardware

The analog interface cards listed in Table 6-3 are used to provide low-density analog PSTN interfaces. VICs are placed in a WAN interface card (WIC) slot (supported on the Cisco 1751 and 1760), in a high-speed WIC (HWIC) slot on the router (supported on the Cisco 2800 and 3800 series), or inside a network module (Cisco 2600, 2800, 3700, and 3800 series) such as the NM-HD-1V, NM-HD-2V, NM-HD-2VE, or NM-HDV2. For high-density analog PSTN interfaces, the NM-HDA (supported on the Cisco 2600, 2811, 2821, 2851, 3700, and 3800 series) or the EVM-HD-8FXS/DID card (supported on the Cisco 2821, 2851, and 3800 series) can be used.

The cards that support multiple signaling systems (such as FXS or DID, and FXO or CAMA) can be software configured on a per-port basis to support one or the other. For example, the VIC2-4FXO card can be configured to support one CAMA and three FXO ports, or two CAMA and two FXO ports.

Configuring Analog Trunks and Stations

All PSTN interfaces are configured as voice ports on the router. When you insert the card into the router, the configuration automatically creates and shows the corresponding voice ports. Directing calls to a voice port is based on the dial plan and is implemented with plain old telephone service (POTS) dial peers.

A full discussion of Cisco IOS dial plan features on POTS and VoIP dial peers is beyond the scope of this book. However, note the use of the 9T directive in the destination-pattern command of the dial peer in Example 6-1, which shows a basic FXO port configuration.

Example 6-1. FXO Port and Dial Peer Configuration

Router#show running-config voice-port 1/0/0 voice-port 1/0/1 dial-peer voice 100 pots description PSTN destination-pattern 9T port 1/0/0 dial-peer voice 100 pots description PSTN destination-pattern 9T port 1/0/1

This command is a quick way of dealing with variable-length PSTN dial plans. The T denotes a timeout. The command destination-pattern 9T instructs the dial peer to match any dialed digits that start with a nine, regardless of how many digits follow. When the timeout expires, the digits are forwarded from the voice port to the PSTN. There are other, more explicit ways to make your destination-pattern commands match calls to the PSTN more exactly, including 9911, 9411, 91T, and 9[2-9].

The dial peers shown in Example 6-1 direct all calls (of a varying number of digits) that start with a nine to the two PSTN FXO trunks, ports 1/0/0 and 1/0/1. If no preference is given on the dial peers and both trunks are free, the Cisco IOS software chooses one of the two trunks at random. You can control the order in which they are chosen by adding a preference command to the dial peer. The dial-peer hunt command offers additional control over the sequence in which dial peers, and therefore voice ports, are chosen.

You can also direct calls to different destinations over different trunks if you want to. This is shown in Example 6-2, where calls to the 408 area code always use voice port 1/0/0, and calls to the 415 area code always use voice port 1/0/1. This way, if you have different local and long-distance PSTN provider connections, you can connect to each independently and direct different types of calls to the correct trunks.

Example 6-2. Different PSTN Numbers to Different Trunks

Router#show running-config voice-port 1/0/0 voice-port 1/0/1 dial-peer voice 100 pots description PSTN destination-pattern 9408....... port 1/0/0 dial-peer voice 101 pots description PSTN destination-pattern 9415....... port 1/0/1

If you require CAMA connectivity to comply with North American emergency calling regulations, you can configure one or more of your FXO ports for CAMA operation. This is shown in Example 6-3, where port 2/0/3 on a VIC2-4FXO card is configured for CAMA signaling.

Example 6-3. CAMA Configuration

Router#show running-config voice-port 2/0/0 voice-port 2/0/1 signal ground-start voice-port 2/0/2 voice-port 2/0/3 signal cama KP-NPD-NXX-XXXX-ST

Analog Trunk Features

With analog FXO interfaces, caller ID information received for an incoming PSTN call is displayed on the IP phones. You can optionally enable the Flash softkey on your IP phones. Pressing the Flash softkey on the IP phone generates a hookflash signal on the FXO port and allows you to exercise PSTN subscriber line services, such as PSTN call waiting and three-way calling. However, Cisco IOS FXO ports do not support PSTN call waiting caller ID display.

You can also set up a direct link between a specific PSTN telephone line and an individual button on an IP phone. This is useful if you want to use PSTN-based voice mail services where a stutter dial tone on the PSTN line indicates that a message is waiting.

As mentioned earlier, in the section "Analog Signaling," FXO interfaces are asymmetric. As such, calls can be disconnected in only one direction in pure FXO operation. The historic reasons for this are beyond the scope of this book. Suffice it to say that today FXO ports are widely used as two-way trunks, and special care must be taken that calls disconnect properly in both directions and do not hang the port. You can use the following Cisco IOS commands on the voice port to facilitate proper call disconnect on FXO ports:

• signal groundstart
• battery-reversal
• supervisory disconnect dualtone
• supervisory disconnect anytone

Which of these methods you should use depends on the complementary features provided by your PSTN CO switch. It also varies based on your geographic location and the technology available in the CO you connect to.

In addition, FXO signaling does not receive dialed digits (that is, DNIS). This means that an incoming call from the PSTN to an FXO port cannot be switched automatically by your Cisco CME system to an extension, because there are no digits from the PSTN to tell Cisco CME where to switch it. You can overcome this shortcoming of FXO signaling by using auto-terminate directives on the FXO voice port to switch the call to a predetermined destination. Commands you can explore include connection plar and connection plar-opx, which you will learn more about in the later section "PSTN Call Switching."

Digital Trunks

Digital trunks can be low-density (for example, BRI with two calls per port) or high-density T1 or E1 ports with 24 or 30 calls per port, respectively. The specific hardware cards offering digital trunk interfaces are discussed next.

Digital Trunk Hardware

The digital interface cards listed in Table 6-4 are used to provide a range of low- to high-density digital PSTN interfaces.

In the general case, a T1 port offers 24 voice channels, and an E1 port offers 30 channels. When using ISDN signaling, where one channel is dedicated to call control signaling (the D channel), a T1 carries 23 voice channels, and an E1 carries 30 voice channels. (An E1 always has a channel dedicated to signaling, no matter what type of protocol is used. With T1 this is not normally the case; using ISDN takes away one of the standard channels.)

You do not have to use the maximum number of channels on these ports, depending on what your PSTN service provider offers. You can configure your Cisco IOS router with any number of channels on the T1 or E1 interface, but it has to be complemented by what is configured on the PSTN CO on the other side.

Fractional T1 service is quite common in North America, where you can subscribe to PSTN T1 service with, for example, only 12 or 16 channels of service (and this service costs less than a full T1 of 24 channels). This service can be either T1 CAS or T1 PRI. Another service is to multiplex your WAN connection (Frame Relay or Point-to-Point Protocol [PPP]) on some channels of the same physical T1 used for your PSTN voice connection. For example, channels 1 to 6 could offer a 384-Kbps PPP WAN connection, and channels 10 to 20 could offer ten channels of PSTN voice service using T1 E&M signaling.

Fractional E1 service is much less common or isn't available at all. Your lower-density PSTN connectivity options in geographies that use E1 connectivity may be multiples of BRI until such time as a full E1 makes sense for your business.

Configuring Digital Trunks

Digital PSTN interfaces are configured in general just like analog interfacesthat is, as voice ports and POTS dial peers on the router to direct calls to the ports. The dial peer control and configuration are exactly the same, regardless of what type of voice port you're using.

T1/E1 ports, however, show up as controllers in your basic configuration (by just inserting the hardware into the router). Unlike an analog interface, the voice port is not created until you add more configuration details to the controller. T1/E1 ports are used for both data and voice access. Until you add specific configuration statements, the router does not know what your intention is with the T1/E1 port. You add a voice configuration to a T1/E1 port by using either the ds0-group or pri-group command. A data T1/E1 port is configured with the channel-group command.

You often see the terms CAS and common channel signaling (CCS) when reading about T1/E1 trunks. CAS generally means that the signaling to control the call uses the same channel (or timeslot) as the call's media path. This is common on T1 interfaces. (It is also called robbed-bit signaling because a few bits out of the 64-kbps channel are "stolen" from the media path to convey call control information, such as on-hook and off-hook.) CCS means that a channel is dedicated to signaling. This channel carries the call control information for all the voice calls (media paths) on that same T1/E1 interface. For example, channel 16 on an E1 is used exclusively for call control and carries the control information for all the other channels (1 to 15 and 17 to 31) on that interface.

Example 6-4 shows a T1 CAS (E&M immediate start) PSTN connection using a ds0-group configuration. In this example, you can see that the second port on the VWIC shows up as controller T1 2/1. This means that the hardware has been detected but no configuration has been done for this port.

Example 6-4. T1 CAS Configuration

Router#show running-config controller T1 2/0 framing esf clock source internal linecode b8zs ds0-group 0 timeslots 1-24 type e&m-immediate-start controller T1 2/1 voice-port 2/0:0 signal immediate dial-peer voice 100 pots description PSTN destination-pattern 9T port 2/0:0

In this example, all 24 channels on the T1 are configured. But you could as easily have stated ds0-group 0 timeslots 1-10 if you agreed with your provider to get only ten channels of PSTN service on this T1 (fractional T1 service). The result of the ds0-group command is that voice port 2/0:0 is created. The POTS dial peer, in this example, looks the same as the one in the FXO example earlier, except that it now points to voice port 2/0:0, which is a T1 port.

If you are using ISDN PRI service to the PSTN, you use the pri-group command to insert a voice configuration on a T1 or E1 controller. Example 6-5 shows a sample configuration for a T1 PRI trunk.

Example 6-5. T1 PRI Configuration

Router#show running-config isdn switch-type primary-5ess controller T1 2/0 framing esf linecode b8zs pri-group timeslots 1-24 interface Serial2/0:23 no ip address isdn switch-type primary-5ess isdn incoming-voice voice voice-port 2/0:23 echo-cancel coverage 64 dial-peer voice 100 pots description PSTN destination-pattern 9T port 2/0:23

Geographic variants of ISDN are controlled by the switch-type setting. A default router setting, seen in Example 6-5 as the first line in the configuration, is specified at the Cisco IOS global level (the isdn switch-type command). This default can be overridden on a per-interface basis by the switch-type statement under the controller. In Example 6-5, both are set to primary-5ess, but they could be different. If they are different, the statement on the controller takes precedence.

The D channel interface (interface Serial 2/0:23) and voice-port (voice-port 2/0:23) commands are automatically created by the insertion of the pri-group command on the controller. The POTS dial peer again looks exactly the same as in Examples 6-1, 6-2, and 6-4. You simply have to adjust the voice port it refers to.

Digital Trunk Features

For PRI/BRI interfaces using ISDN signaling, you can optionally allow the IP phone's full DID name and number to be used as the calling party's identity for outgoing calls. This puts extension-specific information into the PSTN billing records for the call. This can be useful if you want to rely on the PSTN provider's billing information to track the internal origin point of PSTN calls made from your Cisco CME system. Alternatively, you can block IP phone extension-specific information from the outgoing ISDN call and instead substitute the general public phone number for your system.

Generally, PSTN providers do not use name information delivered to the PSTN by a subscriber system. Although the name can be included in the ISDN call setup, the PSTN typically overrides this with the information associated with the subscriber in the PSTN's own databases. You can, however, receive name display information from the PSTN on ISDN trunks, and display this on the IP phones in your business.

All digital trunks provide DID (or DNIS) information. ISDN trunks also provide caller ID delivery. Fractional CAS and PRI are supported on the Cisco IOS routers. If you configure fractional PRI, the D channel for the T1 must be on channel 24 and for E1 on channel 16. This cannot be customized. The voice channels (B channels) can be any subset of the remaining channels.

ISDN channels cannot be customized to be incoming only or outgoing only. However, through creative use of dial peers, you can limit the number of incoming or outgoing calls to and from your business. You just cannot specify the exact channel each call should use. With T1 CAS, you have more granular control, because you can specify separate ds0-groups (up to a ds0-group per channel). Each ds0-group creates a separate voice port that you can control via dial peers as to what calls may reach those channels. Example 6-6 shows a sample configuration for this.

Example 6-6. T1 CAS Configuration with Separate Voice Ports

Router#show running-config controller T1 2/0 framing esf clock source internal linecode b8zs ds0-group 0 timeslots 1-10 type e&m-immediate-start ds0-group 1 timeslots 15-20 type e&m-immediate-start controller T1 2/1 voice-port 2/0:0 signal immediate voice-port 2/0:1 signal immediate dial-peer voice 100 pots description PSTN destination-pattern 9408....... port 2/0:0 dial-peer voice 101 pots description PSTN destination-pattern 9415....... port 2/0:1

The ds0-group 0 timeslots 1-10 command results in voice port 2/0:0, and the ds0-group 1 timeslots 15-20 command creates voice port 2/0:1.

DSP Hardware

Digital signal processor (DSP) technology provides voice compression, echo cancellation, tone generation, and voice packetization functions for servicing voice interfaces and converting the voice for transport over packet networks. To drive a PSTN voice connection, the analog or digital voice ports must have access to a DSP for the call.

Some voice NMs include internal slots into which DSP modules can be plugged, and others have fixed DSP configurations. In some router models, such as the Cisco 1760, 2800, and 3800 series, you can plug DSP cards directly into the router's motherboard.

VWIC cards offer only physical T1/E1 port connections, and VIC cards offer only the physical analog or BRI ports. If a VIC or VWIC card is inserted into a router WIC slot (supported on the Cisco 1751, 1760, 2800, and 3800 series), the DSPs are typically provided by the onboard DSP cards. A VIC or VWIC inserted into an NM typically draws on DSPs resident on the NM itself.

One other variation is to use a VWIC in a WIC slot on the Cisco 2600 or 3700 series platforms, which do not support onboard DSPs. For this configuration, you can use a DSP AIM card such as the AIM-VOICE-30 or the AIM-ATM-VOICE-30 card. (An Advanced Integration Module [AIM] is an internal plug-in module that fits on the router's motherboard.) The AIM-based DSPs cannot drive analog or BRI VIC cards, only T1/E1 VWICs.

DSP cards for motherboard and NM-based slots come in many densities and use various DSP technologies. All are called packet voice/fax DSP module (PVDM) cards.

PSTN Trunks Integrated with or Separate from Cisco CME

In a typical deployment, the PSTN connectivity for your business is integrated into your Cisco CME router. However, you could also use a separate router platform as your PSTN gateway. You may choose to do this because you already have a router that acts as your PSTN gateway in your office or because the slot density on your Cisco CME router is insufficient for the PSTN connectivity your office requires.

For PSTN trunks integrated onto your Cisco CME router, the voice call is switched directly from the POTS interface to the IP phone and is straightforward to configure. Placing the PSTN gateway on a different platform gives you an H.323 (or SIP) call leg between the PSTN gateway and the Cisco CME call controller where the IP phones are managed. This requires POTS dial peers to direct calls to the PSTN interfaces, as shown in the previous configuration examples in this chapter. It also requires H.323 dial peers to direct calls from the PSTN gateway to IP phones, as well as from the IP phones to the PSTN gateway. From an H.323 standpoint, this configuration is similar to connecting two separate Cisco CME systems via an H.323 VoIP interface between them. This is shown in Figure 6-1.

Figure 6-1. Integrated or Separate PSTN Gateways

It is recommended that you deploy Cisco CME with an integrated PSTN gateway, because this results in a much simpler network design and configuration. Another consideration is that if Cisco Unity Express (UE) is used for the automated attendant (AA) or voice mail on your Cisco CME system, the H.323 VoIP leg must be converted to a SIP call leg before the call can successfully terminate on the Cisco UE application. In Cisco CME 3.2 and later, you can do this using the Cisco IOS translation shown in Example 6-7. In Cisco CME 3.1 and earlier, a workaround using loopback-dns can be used, but this is not recommended. It complicates your configuration considerably and has several caveats about particular call flows that are not supported with loopback-dns, such as T.38 fax relay. It is better to upgrade to Cisco CME 3.2.

Example 6-7. H.323 to SIP Translation for Calls into Cisco UE

Router#show running-config voice service voip allow-connections h323 to sip