Voice Port Configuration Guide, Cisco IOS Release 12.4T
Configuring Analog Voice Ports
Configuring Analog Voice Ports
Last Updated: December 14, 2011
Analog voice port interfaces connect routers in packet-based networks to analog two-wire or four-wire analog circuits in telephony networks. Two-wire circuits connect to analog telephone or fax devices, and four-wire circuits connect to PBXs. Connections to the PSTN central office (CO) are typically made with digital interfaces.
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Prerequisites for Configuring Analog Voice Ports
If you are connecting a voice-port interface to a PBX, it is important to understand the PBX's wiring scheme and timing parameters. This information should be available from your PBX vendor or the reference manuals that accompany your PBX.
Information About Analog Voice Hardware
Cisco 880 Series Routers
Beginning with Cisco IOS Release 12.4(15)XZ, the Cisco 880 series fixed router platforms support the implementation of analog (FXS/DID/FXO) and digital (BRI S/T) voice ports. The IAD881B, IAD881F, IAD888B, and IAD888F models support voice interface FXS or BRI. The IAD881F and IAD888F models have four FXS ports and the IAD881B and IAD888B models support two ports for ISDN BRI digital voice interface.
In the IAD881B and IAD888B models, the voice BRI interface presents an ISDN S/T interface to connect either to an NT1 terminating an ISDN telephone network (TE-side) or to a TE user device such as an ISDN telephone or PBX (NT-side). In the IAD881B and IAD888B models, the BRI interface is available as the primary voice interface and is intended to be connected to a PBX (network side trunk). All the voice interfaces are onboard though they are recognized as a 4-port FXS VIC and a 2-port BRI VIC in order to leverage existing voice drivers.
The C881 and C888 SRST models automatically detect a failure occuring in the network and initiate a process to auto-configure the router. This process provides call-processing backup redundancy for the IP and FXS phones and helps to ensure that telephony capabilities stay operational. All the IP or analog phones hanging off of a telecommuter site are controlled by the headquarters office call control (Cisco Unified CallManager or CallManager Express). In case of a WAN failure, the telecommuter router allows all phones to re-register to it in SRST mode and allow all inbound and outbound dialing to be routed off to the PSTN (using back up FXO or BRI port). Upon restoration of WAN connectivity, the system automatically shifts call processing back to the primary Cisco Unified Call Manager cluster.
Cisco 1750 Modular Router
The Cisco 1750 modular router provides VoIP functionality and can carry voice traffic (for example, telephone calls and faxes) over an IP network. To make a voice connection, the router must have a supported voice interface card (VIC) installed. The Cisco 1750 router supports two slots for either WAN interface cards (WICs) or VICs and supports one VIC-only slot. For analog connections, two-port VICs are available to support FXO, FXS, and E&M signaling. VICs provide direct connections to telephone equipment (analog phones, analog fax machines, key systems, or PBXs) or to a PSTN.
Cisco2600 Cisco 3600 and Cisco 3700 Series Routers
The Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series routers are modular, multifunction platforms that combine dial access, routing, LAN-to-LAN services, and multiservice integration of voice, video, and data in the same device.
Voice network modules installed in Cisco 2600 series, Cisco 3600 series, or Cisco 3700 series routers convert telephone voice signals into data packets that can be transmitted over an IP network. The voice network modules have no connectors; VICs installed in the network modules provide connections to the telephone equipment or network. VICs work with existing telephone and fax equipment and are compatible with H.323 standards for audio and video conferencing.
For analog telephone connections, low-density voice/fax network modules that contain either one or two VIC slots are installed in the network module slots. Each VIC is specific to a particular telephone signaling interface (FXS, FXO, or E&M); therefore, the VIC determines the type of signaling on that module.
For more information, refer to the following:
To support analog voice circuits, a Cisco MC3810 must be equipped with an AVM, which supports six analog voice ports. When you install specific signaling modules known as analog personality modules (APMs), the analog voice ports may be equipped for the following signaling types in various combinations: FXS, FXO, and E&M. For FXS, the analog voice ports use an RJ-11 connector interface to connect to analog telephones or fax machines (two-wire) or to a key system (four-wire). For FXO, the analog voice ports use an RJ-11 physical interface to connect to a CO trunk. For E&M connections, the analog voice ports use an RJ-1CX physical interface to connect to an analog PBX (two-wire or four-wire).
Optional high-performance voice compression modules (HCMs) can replace standard voice compression modules (VCMs) to operate according to the voice compression coding algorithm (codec) specified when the Cisco MC3810 concentrator is configured. The HCM2 provides four voice channels at high codec complexity and eight channels at medium complexity. The HCM6 provides 12 voice channels at high complexity and 24 channels at medium complexity. One or two HCMs can be installed in a Cisco MC3810, but an HCM may not be combined with a VCM in one chassis.
For more information, refer to the Cisco MC3810 Multiservice Concentrator Hardware Installation Guide .
Basic Parameters on Analog FXO FXS or E and M Voice Ports
This section describes commands for basic analog voice port configuration.
All the data recommended in the Prerequisites for Configuring Analog Voice Ports should be gathered before you start this procedure.
If you are configuring a Cisco MC3810 that has HCMs, you should also configure the codec complexity by performing the tasks in the Configuring Codec Complexity on the Cisco MC3810.
In addition to the basic voice port parameters described in this section, there are commands that allow voice port configurations to be fine-tuned. In most cases, the default values for fine-tuning commands are sufficient for establishing FXO and FXS voice port configurations. E&M voice ports are more likely to require some configuration. If it is necessary to change some of the voice port values to improve voice quality or to match parameters on proprietary PBXs to which you are connecting, use the commands in this section and also in the "Fine-Tuning Analog and Digital Voice Ports" chapter.
After the voice port has been configured, make sure that the ports are operational by performing the tasks described in the following chapters:
For more information on these and other voice port commands, refer to the Cisco IOS Voice Command Reference.
Codec Complexity for Analog Voice Ports on the Cisco MC3810 with High-Performance Compression Modules
The term codec stands for coder-decoder. A codec is a particular method of transforming analog voice into a digital bit stream (and vice versa) and also refers to the type of compression used. Several different codecs have been developed to perform these functions, and each one is known by the number of the International Telecommunication Union-Telecommunication Standardization Sector (ITU-T) standard in which it is defined. For example, two common codecs are the G.711 and the G.729 codecs. The various codecs use different algorithms to encode analog voice into digital bit-streams and have different bit rates, frame sizes, and coding delays associated with them. The codecs also differ in the type of perceived voice quality they achieve. Specialized hardware and software in the digital signal processors (DSPs) perform codec transformation and compression functions, and different DSPs may offer different selections of codecs.
Select the same type of codec as the one that is used at the other end of the call. For instance, if a call was coded with a G.729 codec, it must be decoded with a G.729 codec. Codec choice is configured in dial peers. For more information, refer to the "Dial Peer Configuration on Voice Gateway Routers" document.
Codec complexity refers to the amount of processing power that a codec compression method requires. The greater the codec complexity, the fewer the calls that the DSP interfaces can handle. Codec complexity is either medium or high. The default is medium. All medium-complexity codecs can also run in high-complexity mode, but fewer (usually half as many) channels are available per DSP. The codec complexity value determines the choice of codecs that are available in the dial peers when the codec command has been configured. For details on the number of calls that can be handled simultaneously using each of the codec standards, refer to the entries for the codecand codec complexity commands in the Cisco IOS Voice Command Reference.
How to Configure Analog Voice Ports
Configuring Basic Parameters on Analog FXO FXS or E and M Voice Ports
Configuring Codec Complexity on the Cisco MC3810
To configure codec complexity for analog voice ports on the Cisco MC3810 using High-Performance Compression Modules (HCMs), use the following commands:
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