Table Of Contents
Configuring Voice over IP for Cisco MC3810 Series Concentrators
Supported Standards, MIBs, and RFCs
Configuring IP Networks for Real-Time Voice Traffic
Configuring Multilink PPP with Interleaving
Multilink PPP Configuration Example
Configuring RTP Header Compression
Enable RTP Header Compression on a Serial Interface
Change the Number of Header Compression Connections
RTP Header Compression Configuration Example
Configuring Dial Peer Digit Manipulation
Optimizing Dial Peer and Network Interface Configurations
Configuring IP Precedence for Dial Peers
Configuring Codec and VAD for Dial Peers
Configuring Codec for a VoIP Dial Peer
Configuring VAD for a VoIP Dial Peer
Configuring Codec Selection Order
Configuring a Voice Class to Define Codec Selection Order
Applying a Voice Class for Codec Selection to a VoIP Dial Peer
Verifying Codec Settings of Dial Peers
Configuring FXO or FXS Voice Ports
Fine-Tuning FXO and FXS Voice Ports
Configuring POTS and VoIP Dial Peers
Enabling VoIP Gateway Functionality
Configuring Gateway Interface Parameters
Linking PBX Users with E&M Trunk Lines
PSTN Gateway Access Using FXO Connection
PSTN Gateway Access Using FXO Connection (PLAR Mode)
Codec Preference Configuration
Configuring Voice over IP for Cisco MC3810 Series Concentrators
Feature Summary
Voice over IP (VoIP) enables a Cisco MC3810 concentrator to carry voice traffic (for example, telephone calls and faxes) over an IP network. Voice over IP is primarily a software feature; however, to support this feature, a Cisco MC3810 must be equipped with a digital voice module (DVM) or an analog voice module (AVM). The Cisco MC3810's LAN/WAN multiservice routing capabilities provide analog and digital (T1/E1) VoIP gateway capabilities for packetized voice traffic.
In Voice over IP, the DSP segments the voice signal into frames, which are then coupled in groups of two and stored in voice packets. These voice packets are transported using IP in compliance with ITU-T specification H.323. Because it is a delay-sensitive application, you need to have a well-engineered network end-to-end to successfully use Voice over IP. Fine-tuning your network to adequately support Voice over IP involves a series of protocols and features geared toward quality of service (QoS). Traffic shaping considerations must be taken into account to ensure the reliability of the voice connection.
Benefits
Voice over IP offers the following benefits:
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Related Documents
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Supported Platform
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Supported Standards, MIBs, and RFCs
This feature supports the following standards and RFCs:
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Prerequisites
The voice enhancements described in this document require the use of Cisco IOS Release 12.0(7)XK or newer.
Configuration Tasks
To configure Voice over IP on the Cisco MC3810 concentrator, you need to complete the following tasks:
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Configure your IP network to support real-time voice traffic. Fine-tuning your network to adequately support VoIP involves a series of protocols and features geared toward quality of service (QoS). To configure your IP network for real-time voice traffic, you need to take into consideration the entire scope of your network, then select and configure the appropriate QoS tool or tools:
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Refer to the "Configuring IP Networks for Real-Time Voice Traffic" section for information about how to select and configure the appropriate QoS tools to optimize voice traffic on your network.
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Use the num-exp command to configure number expansion if your telephone network is configured so that you can reach a destination by dialing only a portion (an extension number) of the full E.164 telephone number. Refer to the "Configuring Number Expansion" section for information about number expansion.
Use the dial-peer voice command to define dial peers and switch to the dial-peer configuration mode. Each dial peer defines the characteristics associated with a call leg. A call leg is a discrete segment of a call connection that lies between two points in the connection. An end-to-end call is comprised of four call legs, two from the perspective of the source access server, and two from the perspective of the destination access server. Dial peers are used to apply attributes to call legs and to identify call origin and destination. There are two different kinds of dial peers:
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Refer to the "Configuring Dial Peers" section for additional information about configuring dial peers and dial-peer characteristics.
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You can use VoIP peers to define characteristics such as IP precedence, CODEC, and VAD. Use the ip precedence command to define IP precedence. Use the codec command to configure specific voice coder rates. Use the vad command to disable voice activation detection and the transmission of silence packets. Refer to the "Optimizing Dial Peer and Network Interface Configurations" section for additional information about optimizing dial-peer characteristics.
You need to configure your Cisco MC3810 concentrator to support voice ports. In general, voice-port commands define the characteristics associated with a particular voice-port signaling type. Voice ports on the Cisco MC3810 concentrator support three basic voice signaling types:
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Under most circumstances, the default voice-port command values are adequate to configure FXO and FXS ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks, E&M ports might need specific voice-port values configured, depending on the specifications of the devices in your telephony network.
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The gateway capability allows a Cisco MC3810 to function as an H.323 endpoint. Therefore, the gateway provides admission control, and address lookup and translation.
Preparing to Configure VoIP
Before you can configure your Cisco MC3810 concentrator to use Voice over IP, you must first:
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After you have analyzed your dial plan and decided how to integrate it into your existing IP network, you are ready to configure your network devices to support Voice over IP.
Configuring IP Networks for Real-Time Voice Traffic
You need to have a well-engineered network end-to-end when running delay-sensitive applications such as VoIP. Fine-tuning your network to adequately support VoIP involves a series of protocols and features geared toward quality of service (QoS). It is beyond the scope of this document to explain the specific details relating to wide-scale QoS deployment. Cisco IOS software provides many tools for enabling QoS on your backbone, such as Random Early Detection (RED), Weighted Random Early Detection (WRED), Fancy queuing (meaning custom, priority, or weighted fair queuing), and IP Precedence. To configure your IP network for real-time voice traffic, you need to take into consideration the entire scope of your network, then select the appropriate QoS tool or tools.
The important thing to remember is that QoS must be configured throughout your network—not just on the Cisco MC3810 concentrator devices running VoIP—to improve voice network performance. Not all QoS techniques are appropriate for all network routers. Edge routers and backbone routers in your network do not necessarily perform the same operations; the QoS tasks they perform might differ as well. To configure your IP network for real-time voice traffic, you need to take into consideration the functions of both edge and backbone routers in your network, then select the appropriate QoS tool or tools.
In general, edge routers perform the following QoS functions:
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In general, backbone routers perform the following QoS functions:
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Scalable QoS solutions require cooperative edge and backbone functions.
Although not mandatory, some QoS tools have been identified as being valuable in fine-tuning your network to support real-time voice traffic. To configure your IP network for QoS using these tools, perform one or more of the following tasks:
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Each of these components is discussed in the following sections.
Configuring Multilink PPP with Interleaving
Multiclass Multilink PPP Interleaving allows large packets to be multilink-encapsulated and fragmented into smaller packets to satisfy the delay requirements of real-time voice traffic; small real-time packets, which are not multilink-encapsulated, are transmitted between fragments of the large packets. The interleaving feature also provides a special transmit queue for the smaller, delay-sensitive packets, enabling them to be transmitted earlier than other flows. Interleaving provides the delay bounds for delay-sensitive voice packets on a slow link that is used for other best-effort traffic.
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In general, Multilink PPP with interleaving is used in conjunction with weighted fair queuing or IP Precedence to ensure voice packet delivery. Use Multilink PPP with interleaving and weighted fair queuing to define how data will be managed; use IP Precedence to give priority to voice packets.
You should configure Multilink PPP if the following conditions exist in your network:
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Multilink PPP support for interleaving can be configured on virtual templates, dialer interfaces, and ISDN BRI or PRI interfaces. To configure interleaving, you need to complete the following tasks:
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To configure Multilink PPP and interleaving on a configured and operational interface or virtual interface template, use the following commands in interface mode:
For more information about Multilink PPP, refer to the "Configuring Media-Independent PPP and Multilink PPP" chapter in the Dial Solutions Configuration Guide.
Multilink PPP Configuration Example
The following example defines a virtual interface template that enables Multilink PPP with interleaving and a maximum real-time traffic delay of 20 milliseconds, and then applies that virtual template to the Multilink PPP bundle:
interface virtual-template 1ppp multilinkencapsulated pppppp multilink interleaveppp multilink fragment-delay 20ip rtp priority 16384 16383 25multilink virtual-template 1Configuring RTP Header Compression
Real-Time Transport Protocol (RTP) is used for carrying packetized audio traffic over an IP network. RTP header compression compresses the IP/UDP/RTP header in an RTP data packet from 40 bytes to approximately 2 to 4 bytes (most of the time), as shown in .
This compression feature is beneficial if you are running Voice over IP over slow links. Enabling compression on both ends of a low-bandwidth serial link can greatly reduce the network overhead if there is a lot of RTP traffic on that slow link.
Typically, an RTP packet has a payload of approximately 20 to 160 bytes for audio applications that use compressed payloads. RTP header compression is especially beneficial when the RTP payload size is small (for example, compressed audio payloads between 20 and 50 bytes).
Figure 1 RTP Header Compression
You should configure RTP header compression if the following conditions exist in your network:
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Perform the following tasks to configure RTP header compression for Voice over IP. The first task is required; the second task is optional.
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Enable RTP Header Compression on a Serial Interface
To use RTP header compression, you need to enable compression on both ends of a serial connection. To enable RTP header compression, use the following command in interface configuration mode:
router(config-if)# ip rtp header-compression [passive]
Enable RTP header compression.
If you include the passive keyword, the software compresses outgoing RTP packets only if incoming RTP packets on the same interface are compressed. If you use the command without the passive keyword, the software compresses all RTP traffic.
Change the Number of Header Compression Connections
By default, the software supports a total of 32 RTP header compression connections on an interface. To specify a different number of RTP header compression connections, use the following command in interface configuration mode:
router(config-if)# ip rtp compression connections number
Specify the total number of RTP header compression connections supported on an interface.
RTP Header Compression Configuration Example
The following example enables RTP header compression for a serial interface:
interface 0ip rtp header-compressionencapsulation pppip rtp compression-connections 25For more information about RTP header compression, see the "Configuring IP Multicast Routing" chapter of the Network Protocols Configuration Guide, Part 1.
Configuring IP RTP Priority
IP RTP Priority provides a strict priority queueing scheme for delay-sensitive data such as voice. Voice traffic can be identified by its Real-Time Transport Protocol (RTP) port numbers and classified into a priority queue configured by the ip rtp priority command. The result is that voice is serviced as strict priority in preference to other nonvoice traffic.
This feature allows you to specify a range of User Datagram Protocol (UDP)/RTP ports whose voice traffic is guaranteed strict priority service over any other queues or classes using the same output interface. Strict priority means that if packets exist in the priority queue, they are dequeued and sent first—that is, before packets in other queues are dequeued.
The IP RTP Priority feature does not require that you know the port of a voice call. Rather, the feature gives you the ability to identify a range of ports whose traffic is put into the priority queue. Moreover, you can specify the entire voice port range—16384 to 32767—to ensure that all voice traffic is given strict priority service. IP RTP Priority is especially useful on slow-speed links whose speed is less than 1.544 Mbps.
This feature can be used in conjunction with Weighted Fair Queueing (WFQ) on the same outgoing interface.Traffic matching the range of ports specified for the priority queue is guaranteed strict priority over other WFQ flows; voice packets in the priority queue are always serviced first.
When used in conjunction with WFQ, the ip rtp priority command provides strict priority to voice, and WFQ scheduling is applied to the remaining queues.
Because voice packets are small in size and the interface also can have large packets going out, the Link Fragmentation and Interleaving (LFI) feature should also be configured on lower speed interfaces. When you enable LFI, the large data packets are broken up so that the small voice packets can be interleaved between the data fragments that make up a large data packet. LFI prevents a voice packet from needing to wait until a large packet is sent. Instead, the voice packet can be sent in a shorter amount of time.
For more information about the IP RTP Priority feature, see the IP RTP Priority Cisco IOS Release 12.0(5)T online document.
To reserve a strict priority queue for a set of RTP packet flows belonging to a range of UDP destination ports, use the following command in interface configuration mode:
Configuring Number Expansion
This section describes how to use the num-exp command to expand a set of dialed digits, such as an extension number, into a destination pattern representing a complete telephone number for Voice over IP on Cisco MC3810 concentrators.
Enter the following command in global configuration mode for each extension number to be expanded into a destination pattern.
Configuring Dial Peers
This section describes how to use new commands defining dial-peer operation for Voice over IP on Cisco MC3810 series concentrators.
Configure POTS Dial Peers
POTS dial peers enable incoming calls to be received by a particular telephony device. To configure a POTS peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its telephone number(s), and associate it with a voice port through which calls will be established. Under most circumstances, the default values for the remaining dial-peer configuration commands will be sufficient to establish connections.
To enter dial-peer configuration mode (and select POTS as the method of voice-related encapsulation), use the following command in global configuration mode:
router(config)# dial-peer voice number pots
Enter the dial-peer configuration mode to configure a POTS peer.
The number value of the dial-peer voice pots command is a tag that uniquely identifies the dial peer. (This number has local significance only.) The tag value identifies the dial peer and must be unique on the router. Do not duplicate a specific tag number.
To configure the identified POTS peer, use the following commands in dial-peer configuration mode:
To configure direct inward dial (DID) for a particular POTS dial peer, use the following commands beginning in global configuration mode:
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For additional POTS dial-peer configuration options, refer to the "Voice-Related Commands" section of the Cisco IOS 12.0 Voice, Video, and Home Applications Command Reference.
Configure VoIP Peers
VoIP peers enable outgoing calls to be made from a particular telephony device. To configure a VoIP peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its destination telephone number and destination IP address. As with POTS peers, under most circumstances, the default values for the remaining dial-peer configuration commands will be adequate to establish connections.
To enter the dial-peer configuration mode (and select VoIP as the method of voice-related encapsulation), use the following command in global configuration mode:
router(config)#dial-peer voice number voip
Enter the dial-peer configuration mode to configure a VoIP peer.
The number value of the dial-peer voice voip command is a tag that uniquely identifies the dial peer.
To configure the identified VoIP peer, use the following commands in dial-peer configuration mode:
For additional VoIP dial-peer configuration options, refer to the "Voice-Related Commands" section of the Cisco IOS 12.0 Voice, Video, and Home Applications Command Reference. For examples of how to configure dial peers, refer to the section, "Voice over IP Configuration Examples."
Validation Tips
You can check the validity of your dial-peer configuration by performing the following tasks:
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Troubleshooting Tips
If you are having trouble connecting a call and you suspect the problem is associated with dial-peer configuration, you can try to resolve the problem by performing the following tasks:
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Configuring Dial Peer Hunting
After you have configured dial peers, you can configure how the router or concentrator performs dial-peer hunting functions. To configure dial-peer hunting behavior, perform the following steps beginning in global configuration mode.
If using dial peer hunting, there may be situations in which you want to disable dial-peer hunting on a specific dial peer. To disable dial-peer hunting on a dial peer, use the following commands beginning in global configuration mode:
To reenable dial-peer hunting on a dial peer, enter the no huntstop command.
Configuring Dial Peer Digit Manipulation
After you have configured dial peers, you can configure the dial-peer digit manipulation. To configure dial-peer digit manipulation, perform one or more of the following steps beginning in dial-peer configuration mode.
Optimizing Dial Peer and Network Interface Configurations
Depending on how you have configured your network interfaces, you might need to configure additional VoIP dial-peer parameters. This section describes the following topics:
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Configuring IP Precedence for Dial Peers
If you want to give real-time voice traffic a higher priority than other network traffic, you can weight the voice data traffic associated with a particular VoIP dial peer by using IP Precedence. IP Precedence provides no admission control.
To give real-time voice traffic precedence over other IP network traffic, use the following commands, beginning in global configuration mode:
In IP Precedence, the numbers 1 through 5 identify classes for IP flows; the numbers 6 through 7 are used for network and backbone routing and updates.
For example, to ensure that voice traffic associated with VoIP dial peer 103 is given a higher priority than other IP network traffic, enter the following:
dial-peer voice 103 voipip precedence 5In this example, when an IP call leg is associated with VoIP dial peer 103, all packets transmitted to the IP network via this dial peer will have their precedence bits set to 5. If the networks receiving these packets have been configured to recognize precedence bits, the packets will be given priority over packets with a lower configured precedence value.
Configuring Codec and VAD for Dial Peers
Coder-decoder (codec) and voice activity detection (VAD) for a dial peer determine how much bandwidth the voice session uses. Codec typically is used to transform analog signals into a digital bit stream and digital signals back into analog signals—in this case, it specifies the voice coder rate of speech for a dial peer. VAD is used to disable the transmission of silence packets.
Configuring Codec for a VoIP Dial Peer
To specify a voice coder rate for a selected VoIP peer, use the following commands beginning in global configuration mode:
The default for the codec command is g729r8; normally the default configuration for this command is the most desirable. If, however, you are operating on a high bandwidth network and voice quality is of the highest importance, you should configure the codec command for g711alaw or ulaw. Using this value will result in better voice quality, but it will also require higher bandwidth requirements for voice.
For example, to specify a codec rate of G.711a-law for VoIP dial peer 108, enter the following:
dial-peer voice 108 voipdestination-pattern +14085551234codec g711alawsession target ipv4:10.0.0.8Configuring VAD for a VoIP Dial Peer
To disable the transmission of silence packets for a selected VoIP peer, use the following commands beginning in global configuration mode:
The default for the vad command is enabled; normally the default configuration for this command is the most desirable. If you are operating on a high bandwidth network and voice quality is of the highest importance, you should disable vad. Using this value will result in better voice quality, but it will also require higher bandwidth requirements for voice.
For example, to enable VAD for VoIP dial peer 108, enter the following:
dial-peer voice 108 voipdestination-pattern +14085551234vadsession target ipv4:10.0.0.8Configuring Codec Selection Order
To configure codec selection order, perform the following tasks:
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Configuring a Voice Class to Define Codec Selection Order
You can define a voice class in which you configure a selection order for codecs, and then map the voice class to a VoIP dial peer.
To configure a voice class in which you can define the order of preference in which a router selects a codec when it negotiates with a far-end router, enter the following commands beginning in global configuration mode:
Applying a Voice Class for Codec Selection to a VoIP Dial Peer
After you have created the voice class, assign it to a VoIP dial peer. You cannot assign voice-class codec attributes to POTS dial peers.
To apply voice-class signaling attributes to a VoIP dial peer, complete the following steps beginning in global configuration mode:
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Define a VoIP dial peer and enter dial-peer configuration mode. All subsequent commands that you enter in dial-peer voice mode before you exit will apply to this dial peer.
The tag is a number that identifies the dial peer and must be unique on the router. Do not assign duplicate tag numbers.
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Assign to the dial peer the voice class that you created in the "Configuring a Voice Class to Define Codec Selection Order" section.
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Verifying Codec Settings of Dial Peers
To display the codec voice-classes assigned to VoIP dial peers, enter the show running-config command.
The following example shows exerpts from the show running-config command output, where three codec voice classes (10, 20 and 30) have been applied to three VoIP dial peers (101, 102 and 102):
router# show running-configBuilding configuration...Current configuration:!version 12.0...voice class codec 10codec preference 1 g711alawcodec preference 2 g711ulaw bytes 80codec preference 3 g726r16 bytes 120!voice class codec 20codec preference 1 g726r24 bytes 90codec preference 2 g726r32 bytes 120!voice class codec 30codec preference 1 g729ar8codec preference 2 g726r16codec preference 3 g726r32!...dial-peer voice 101 voipvoice-class codec 10!dial-peer voice 102 voipvoice-class codec 20!dial-peer voice 103 voipvoice-class codec 30!line con 0transport input noneline aux 0line 2 3line vty 0 4password #1writerlogin!endConfiguring Voice Ports
This section describes how to configure voice ports for Voice over IP (VoIP) on Cisco MC3810 series concentrators.
Perform the following tasks, as applicable, to configure voice ports:
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Configuring FXO or FXS Voice Ports
Under most circumstances the default values are adequate for FXO and FXS voice ports.
Task List
If you need to change the default configuration for these voice ports, perform the following tasks:
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Configuration Procedure
To configure FXO and FXS voice ports, enter the following commands, beginning in global configuration mode. Commands apply to both analog and digital voice ports unless otherwise indicated.
Validation Tips
You can check the validity of your voice-port configuration by performing the following tasks:
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Troubleshooting Tips
If you are having trouble connecting a call and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:
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Fine-Tuning FXO and FXS Voice Ports
Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation. The commands for these parameters are referred to as voice-port tuning commands.
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Task List
To fine tune FXO and FXS voice ports, perform the following tasks:
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Voice-Port Tuning Procedure
To fine-tune FXO and FXS voice ports, perform the following optional steps, beginning in global configuration mode. Commands apply to both analog and digital voice ports unless otherwise indicated.
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Configuring E&M Voice Ports
The default E&M voice-port parameters will probably not be sufficient to enable voice transmission over your network. Configuration parameters depend on the PBX to which the voice port is connected.
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Task List
To configure E&M voice ports, perform the following tasks:
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Configuration Procedure
To configure E&M voice ports, enter the following commands beginning in global configuration mode. Commands apply to both analog and digital voice ports unless otherwise indicated.
Validation Tips
You can check the validity of your voice-port configuration by performing the following tasks:
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Troubleshooting Tips
If you are having trouble connecting a call and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:
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Fine-Tuning E&M Voice Ports
Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation. The commands for these parameters are referred to as voice-port tuning commands.
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Task List
To fine tune E&M voice ports, perform the following tasks:
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Voice-Port Tuning Procedure
To fine-tune E&M voice ports, perform the following steps, beginning in privileged EXEC mode. Commands apply to both analog and digital voice ports unless otherwise indicated.
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Activating the Voice Port
After you have configured the voice port, you need to activate the voice port to bring it online. Cisco recommends that you cycle the port—shut the port down and then bring it online again.
To activate a voice port, enter the following command in voice-port configuration mode:
To cycle a voice port, enter the following commands in voice-port configuration mode:
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Configuring the H.323 Gateway
In this release, basic gateway Registration, Admission, and Status (RAS) protocol capability is extended to the Cisco MC3810. Other features, such as authentication, authorization, and accounting (AAA) enhancements for security and accounting services, interactive voice response (IVR), Integrated Services Digital Network (ISDN) redirect number support, and rotary call pattern support, will be offered in future Cisco IOS releases.
To configure the H.323 Gateway, you need to perform the following tasks
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Configuring POTS and VoIP Dial Peers
The first step in configuring the H.323 gateway is to define the applicable POTS and VoIP dial peers. The POTS dial peer informs the system which voice port to direct incoming VoIP calls. The VoIP dial peer defines how to direct calls that originate from a local voice port into the VoIP cloud to the session target. The session target command indicates the address of the remote gateway where the call is terminated. There are several different ways to define the destination gateway address: by statically configuring the IP address of the gateway, by defining the DNS of the gateway, or by using RAS. If you use RAS, that gateway determines the destination target by querying the RAS gatekeeper. See the "Configuring Dial Peers" section to define dial peers for VoIP.
Enabling VoIP Gateway Functionality
Enable VoIP gateway functionality by using the gateway command.
To enable gateway functionality, use the following commands:
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Enter global configuration mode.
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router(config)# gateway
Enable the VoIP gateway.
Configuring Gateway Interface Parameters
The next step in configuring an H.323 gateway is to configure the gateway interface parameters. First define which interface will be presented to the VoIP network as this gateway's H.323 interface. Only one interface is allowed to be the gateway interface. You can select either the interface that is connected to the gatekeeper or a loopback interface. The interface that is connected to the gatekeeper is usually a LAN interface (for example, Fast Ethernet, Ethernet, FDDI, or Token Ring).
After you define the gateway interface, configure the gateway to discover the gatekeeper either through multicasting or by directing it to a specific host. Then configure the gateway's H.323 identification number and any technology prefixes that this gateway should register with the gatekeeper.
To define the interface to be used as the H.323 gateway interface and configure the H.323 gateway interface parameters, use the following commands, beginning in global configuration mode:
Configuration Example
The actual Voice over IP configuration procedure you complete depends on the actual topology of your voice network. The following configuration examples should give you a starting point. Of course, these configuration examples would need to be customized to reflect your network topology.
Configuration examples are supplied for the following scenarios:
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These examples are described in the following sections. The following examples use the term "router" to generically describe Cisco routers and concentrators.
Linking PBX Users with E&M Trunk Lines
The following example shows how to configure Voice over IP to link PBX users with E&M trunk lines.
In this example, a company wants to connect two offices: one in San Jose, California and the other in Salt Lake City, Utah. Each office has an internal telephone network using PBX, connected to the voice network by an E&M interface. Both the Salt Lake City and the San Jose offices are using E&M Port Type II, with four-wire operation and ImmediateStart signaling. Each E&M interface connects to the router using two voice interface connections. Users in San Jose dial "8-569" and then the extension number to reach a destination in Salt Lake City. Users in Salt Lake City dial "4-527" and then the extension number to reach a destination in San Jose.
illustrates the topology of this connection example.
Figure 2 Linking PBX Users with E&M Trunk Lines Example
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Configuration for Router SJ
hostname sanjose!Configure pots dial peer 1dial-peer voice 1 potsdestination-pattern 555....port 1/0/0!Configure pots dial peer 2dial-peer voice 2 potsdestination-pattern 555....port 1/0/1!Configure voip dial peer 3dial-peer voice 3 voipdestination-pattern 119....session target ipv4:172.16.65.182!Configure the E&M interfacevoice-port 1/0/0signal immediateoperation 4-wiretype 2voice-port 1/0/1signal immediateoperation 4-wiretype 2!Configure the serial interfaceinterface serial 0/0description serial interface type dce (provides clock)clock rate 2000000ip address 172.16.1.123no shutdownConfiguration for Router SLC
hostname saltlake!Configure pots dial peer 1dial-peer voice 1 potsdestination-pattern 119....port 1/0/0!Configure pots dial peer 2dial-peer voice 2 potsdestination-pattern 119....port 1/0/1!Configure voip dial peer 3dial-peer voice 3 voipdestination-pattern 555....session target ipv4:172.16.1.123!Configure the E&M interfacevoice-port 1/0/0signal immediateoperation 4-wiretype 2voice-port 1/0/0signal immediateoperation 4-wiretype 2!Configure the serial interfaceinterface serial 0/0description serial interface type dteip address 172.16.65.182no shutdown
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PSTN Gateway Access Using FXO Connection
The following example shows how to configure Voice over IP to link users with the PSTN gateway using an FXO connection.
In this example, users connected to Router SJ in San Jose, California can reach PSTN users in Salt Lake City, Utah via Router SLC. Router SLC in Salt Lake City is connected directly to the PSTN through an FXO interface.
illustrates the topology of this connection example.
Figure 3 PSTN Gateway Access Using FXO Connection Example
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Configuration for Router SJ
! Configure pots dial peer 1dial-peer voice 1 potsdestination-pattern +14085554000port 1/0/0! Configure voip dial peer 2dial-peer voice 2 voipdestination-pattern 9...........session target ipv4:172.16.65.182! Configure the serial interfaceinterface serial 0/0clock rate 2000000ip address 172.16.1.123no shutdownConfiguration for Router SLC
! Configure pots dial peer 1dial-peer voice 1 potsdestination-pattern 9...........port 1/0/0! Configure voip dial peer 2dial-peer voice 2 voipdestination-pattern +14085554000session target ipv4:172.16.1.123! Configure serial interfaceinterface serial 0/0ip address 172.16.65.182no shutdownPSTN Gateway Access Using FXO Connection (PLAR Mode)
The following example shows how to configure Voice over IP to link users with the PSTN gateway using an FXO connection (PLAR mode).
In this example, PSTN users in Salt Lake City, Utah, can dial a local number and establish a private line connection in a remote location. As in the previous example, Router SLC in Salt Lake City is connected directly to the PSTN through an FXO interface.
illustrates the topology of this connection example.
Figure 4 PSTN Gateway Access Using FXO Connection (PLAR Mode)
Note
Configuration for Router SJ
! Configure pots dial peer 1dial-peer voice 1 potsdestination-pattern +14085554000port 1/0/0! Configure voip dial peer 2dial-peer voice 2 voipdestination-pattern 9...........session target ipv4:172.16.65.182! Configure the serial interfaceinterface serial 0/0clock rate 2000000ip address 172.16.1.123no shutdownConfiguration for Router SLC
! Configure pots dial peer 1dial-peer voice 1 potsdestination-pattern 9...........port 1/0/0! Configure voip dial peer 2dial-peer voice 2 voipdestination-pattern +14085554000session target ipv4:172.16.1.123! Configure the voice-portvoice-port 1/0/0connection plar 14085554000! Configure the serial interfaceinterface serial 0/0ip address 172.16.65.182no shutdownCodec Preference Configuration
The following example enters voice class codec configuration mode, creates voice class 10, and defines a preference list of 12 codecs:
router(config)# voice class codec 10router(config-class)# codec preference 1 g711alawrouter(config-class)# codec preference 2 g711ulaw bytes 80router(config-class)# codec preference 3 g723ar53router(config-class)# codec preference 4 g723ar63 bytes 144router(config-class)# codec preference 5 g723r53router(config-class)# codec preference 6 g723r63 bytes 120router(config-class)# codec preference 7 g726r16router(config-class)# codec preference 8 g726r24router(config-class)# codec preference 9 g726r32 bytes 80router(config-class)# codec preference 10 g728router(config-class)# codec preference 11 g729br8router(config-class)# codec preference 12 g729r8 bytes 50router(config-class)# exitrouter(config-class)# exitrouter(config)#The following example assigns a voice class 10 to a VoIP dial peer:
router(config)# dial-peer voice 25 voiprouter(config-dial-peer)# voice-class codec 10Command Reference
This section documents new or modified commands. Modified commands are indicated by an asterisk (*). All other commands used on these platforms are documented in the Cisco IOS Release 12.0 command reference publications.
•
•
•
codec preference
To define the order of preference in which network dial peers select codecs, use the codec preference voice-class configuration command. Enter the no form of this command to restore the default order of preference.
codec preference priority codec bytes payload-size
no codec preferenceSyntax Description
priority
The order of selection preference you assign to a codec. The valid range is 1 to 12, where 1 is the highest priority.
codec
Codec options.
Note
g711alaw—G.711 A Law 64000 bps
g711ulaw—G.711 u Law 64000 bps
g723ar53—*G.723.1 Annex A 5300 bps
g723ar63—*G.723.1 Annex A 6300 bps
g723r53— *G.723.1 5300 bps
g723r63—*G.723.1 6300 bps
g726r16—G.726 16000 bps
g726r24— G.726 24000 bps
g726r32—G.726 32000 bps
g728—*G.728 16000 bps
g729abr8—*G.729 Annex A and Annex B 8000 bps
g729ar8—G.729 Annex A 8000 bps
g729br8—*G.729 Annex B 8000 bps
g729r8—G.729 8000 bps
bytes
(Optional) The voice payload for each frame.
payload-size
(Optional) Number of bytes you specify as the voice payload of each frame. Values depend on the codec type and the packet voice protocol. See Table 1 for valid entries and default values.
Defaults
If no codec is specified, dial peers are configured for g729r8 and the voice payload is as shown in Table 1 for G.729r8.
If a codec is specified without the bytes keyword, the voice payload is as shown in Table 1.
Command Modes
Voice class configuration
Command History
Usage Guidelines
The routers at opposite ends of the WAN may have to negotiate the codec selection for the network dial peers. The codec preference command specifies the order of preference for selecting a negotiated codec for the connection. Table 1 describes the voice payload options and default values for the codecs and packet voice protocols.
Examples
The following example shows how to create a voice class and specify a codec selection preference for the voice class starting from global configuration mode:
router(config)# voice class codec 10router(config-class)# codec preference 1 g711alawrouter(config-class)# codec preference 2 g711ulaw bytes 80router(config-class)# codec preference 3 g723ar53router(config-class)# codec preference 4 g723ar63 bytes 144router(config-class)# codec preference 5 g723r53router(config-class)# codec preference 6 g723r63 bytes 120router(config-class)# codec preference 7 g726r16router(config-class)# codec preference 8 g726r24router(config-class)# codec preference 9 g726r32 bytes 80router(config-class)# codec preference 10 g728router(config-class)# codec preference 11 g729br8router(config-class)# codec preference 12 g729r8 bytes 50router(config-class)# exitrouter(config)# exitrouter)#Related Commands
connection
To specify a connection mode for a voice port, use the connection voice-port configuration command. Use the no form of this command to disable the selected connection mode.
connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]}
no connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]}Syntax Description
Defaults
No connection mode is specified.
Command Mode
Voice-port configuration
Command History
Usage Guidelines
Use this command to specify a connection mode for a specific interface. For example, use the connection plar command to specify a PLAR interface. The string you configure for this command is used as the called number for all incoming calls over this connection. The destination peer is determined by the called number.
Use the connection trunk command to specify a straight tie-line connection to a PBX. You can use the connection trunk command for E&M-to-E&M trunks, FXO-to-FXS trunks, and FXS-to-FXS trunks. Signaling will be transported for E&M-to-E&M trunks and FXO-to-FXS trunks; signaling will not be transported for FXS-to-FXS trunks.
If you desire one of the devices in a static trunk connection to act as slave and receive calls only, use the answer-mode option with the connection trunk command when configuring that device.
Note
The connection tie-line command is used on the Cisco router when a dial plan requires that additional digits be added in front of any digits dialed by the PBX, and that the combined set of digits be used to route the call via the dial-peers and into the network. The operation is similar to the connection plar command operation, but in this case the tie-line port also waits to collect digits from the PBX. The tie-line digits are also automatically stripped by a terminating port.
If the connection command is not configured, the standard session application outputs a dial tone when the interface goes off-hook until enough digits are collected to match a dial-peer and complete the call.
Examples
The following example selects PLAR as the connection mode on a Cisco 3600, with a destination telephone number of 555-9262:
router(config)# voice-port 1/0/0router(config-voiceport)# connection plar 5559262The following example selects tie-line as the connection mode on a Cisco MC3810, with a destination telephone number of 555-9262:
router(config)# voice-port 1/1router(config-voiceport)# connection tie-line 5559262The following example specifies a PLAR off-premises extension connection on a Cisco 3600, with a destination telephone number of 555-9262:
router(config)# voice-port 1/0/0router(config-voiceport)# connection plar-opx 5559262The following example configures a Cisco 3600 series router for a trunk connection and specifies that it will establish the trunk only when it receives an incoming call:
router(config)# voice-port 1/0/0router(config-voiceport)# connection trunk 5559262 answer-modeRelated Commands
dial-peer hunt
To specify a hunt selection order for dial-peers, use the dial-peer hunt dial-peer configuration command. Use the no form of this command to restore the default selection order.
dial-peer hunt hunt-order-number
no dial-peer huntSyntax Description
Defaults
The default is longest match in phone number, explicit preference, random selection (hunt order number 0).
Command Mode
Global configuration
Command History
12.0(7)XK
This command was first introduced and was first supported on the Cisco 2600 and 3600 Series routers and on the Cisco MC3810 multiservice access concentrator.
Usage Guidelines
Use the dial-peer hunt dial-peer configuration command if you have configured hunt groups. "Longest match in phone number" refers to the destination pattern that matches the greatest number of the dialed digits. "Explicit preference" refers to the preference setting in the dial-peer configuration. "Least recent use" refers to the destination pattern that has waited the longest since being selected. "Random selection" weights all of the destination patterns equally in a random selection mode.
Example
The following example configures the dial peers to hunt in the following order: (1) longest match in phone number, (2) explicit preference, (3) random selection.
configure terminaldial-peer hunt 0Related Commands
dial-peer terminator
To change the character used as a terminator for variable length dialed numbers, use the dial-peer terminator global configuration command. Use the no form of this command to restore the default terminating character.
dial-peer terminator character
no dial-peer terminatorSyntax Description
character
Designates the terminating character for a variable-length dialed number. Valid numbers and characters are #, *, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, a, b, c, and d. The default is #.
Defaults
The default terminating character is #.
Command Mode
Global configuration
Command History
12.0
This command was introduced.
12.0(7)XK
Usage was restricted to variable-length dialed numbers.
Usage Guidelines
There are certain areas in the world (for example, in certain European countries) where telephone numbers can vary in length. When a dialed-number string has been identified as a variable length dialed-number, the system does not place a call until the configured value for the timeouts interdigits command has expired, or until the caller dials the terminating character. Use the dial-peer terminator global configuration command to change the terminating character.
Example
The following example specifies "9" as the terminating character for variable-length dialed numbers:
configure terminaldial-peer terminator 9#Related Commands
dial-peer voice
To enter dial-peer configuration mode and specify the method of voice encapsulation, use the dial-peer voice global configuration command. Use the no form of this command to disable the selected encapsulation mode.
For the Cisco 2600 series:
dial-peer voice tag {pots | voip | vofr}
no dial-peer voice tagFor the Cisco 3600 series:
dial-peer voice tag {pots | voip | voatm | vofr }
no dial-peer voice tagFor the Cisco MC3810 series:
dial-peer voice tag {pots | voip | voatm | vofr }
no dial-peer voice tagSyntax Description
Defaults
No default behavior or values.
Command Mode
Global configuration
Command History
Usage Guidelines
Use the dial-peer voice global configuration command to switch to the dial-peer configuration mode from the global configuration mode. Use the exit command to exit the dial-peer configuration mode and return to the global configuration mode.
Example
The following example accesses dial-peer configuration mode and configures a POTS peer identified as dial peer 10:
configure terminaldial-peer voice 10 potsRelated Commands
ds0-group
To specify the DS0 timeslots that make up a logical voice port on a T1 or E1 controller, and to specify the signaling type, use the ds0-group controller configuration command. Use the no form of the command to remove the DS0 group and signaling setting.
ds0-group ds0-group-no timeslots timeslot-list type signal-type
no ds0-group ds0-group-no
Syntax Description
Default
No DS0 group is defined.
Command Mode
Controller configuration
Command History
Usage Guidelines
The ds0-group command automatically creates a logical voice port that is numbered as follows:
Cisco 2600 and 3600 series:
slot/port:ds0-group-no.
Cisco MC3810:
slot:ds0-group-no
On the Cisco MC3810, the slot number is the controller number. Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
On the Cisco MC3810 when configured for transparent CCS, the channel type configured as the ext-sig-master is considered the master side of the permanent virtual circuit (PVC) connection which is responsible for establishing the PVC connection. After the master channel is configured, a fixed timer of 30 seconds starts. The voice-signaling driver then generates an off-hook signal on the master voice channel after the timer expires. The call is treated as a regular call, and the master channel does not hang up after the connection is made. If the call does not go through, or if the T1/E1 trunk is down, the 30-second timer on the master channel side restarts. A new off-hook signal is then generated at the master channel side after the timer expires.
Examples
The following example configures ranges of T1 controller timeslots for FXS ground-start and FXO loop-start signaling on a Cisco 2600 or 3600 Series router:
router(config)# controller T1 1/0router(config-controller)# framing esfrouter(config-controller)# linecode b8zsrouter(config-controller)# ds0-group 1 timeslot 1-10 type fxs-ground-startrouter(config-controller)# ds0-group 2 timeslot 11-24 type fxo-loop-startThe following example configures DS0 groups 1 and 2 on controller T1 1 on the Cisco MC3810 to support transparent CCS:
router(config)# controller T1 1 router(config-controller)# mode ccs cross-connectrouter(config-controller)# ds0-group 1 timeslot 1-10 type ext-sig-masterrouter(config-controller)# ds0-group 2 timeslot 11-24 type ext-sig-slaveRelated Command
dtmf-relay
Use the dtmf-relay command to specify how an H.323 gateway relays DTMF tones through an IP network. Options allow the gateway to forward tones "out-of-band", or separate from the voice stream. The no form of this command removes all signaling options and transmits the DTMF tones as part of the audio stream.
dtmf-relay [cisco-rtp] [h245-signal] [h245-alphanumeric]
no dtmf-relaySyntax Description
Default
DTMF tones are sent "inband", or left in the audio stream, unless you use this command.
Command Mode
EXEC
Command History
Usage Guidelines
The dtmf-relay command determines the outgoing format of relayed DTMF tones. The gateway automatically accepts all formats.
The gateway only sends DTMF tones in the format you specify if the remote device supports it. If the remote device supports multiple formats, the gateway chooses the format based on the following priority:
1
2
3
4
Note
Note
Example
The following are two examples of the dtmf-relay command:
•
Router# configure terminalRouter(config)# dial-peer voice 103 voipRouter(config-dial-peer)# dtmf-relay cisco-rtp h245-signalRouter(config-dial-peer)# endRouter#•
Router# configure terminalRouter(config)# dial-peer voice 103 voipRouter(config-dial-peer)# no dtmf-relayRouter(config-dial-peer)# endRelated Commands
dial-peer
Switch to the voice-port configuration mode form the global configuration mode.
forward-digits
To specify which digits to forward for voice calls, use the forward-digits dial-peer configuration command. If the no form of this command is entered, any digits not matching the destination-pattern are not forwarded. Use the default form of this command to restore the default state.
forward-digits {num-digit | all | extra}
no forward-digits
default forward-digitsSyntax Description
Defaults
Dialed digits not matching the destination-pattern are forwarded.
Command Mode
Dial-peer configuration
Command History
Usage Guidelines
This command applies only to POTS dial peers.
Forwarded digits are always right-justified so that extra leading digits are stripped.
The destination pattern includes both explicit digits and wildcards, if present.
Use the default form of this command if a non-default digit-forwarding scheme was entered previously, and you wish to restore the default.
For QSIG ISDN connections, entering forward-digits all implies that all of the digits of the called party number are sent to the ISDN connection. When you enter forward-digits num-digit and enter a number from 1 to 32, the number of digits specified (right justified) of the called part number are sent to the ISDN connection.
Examples
The following example forwards all of the digits in the destination pattern of a POTS dial peer:
dial-peer voice 1 pots destination-pattern 8... forward-digits allThe following example forwards four of the digits in the destination pattern of a POTS dial peer:
dial-peer voice 1 pots destination-pattern 555.... forward-digits 4The following example forwards the extra right-justified digits that exceed the length of the destination pattern of a POTS dial peer:
dial-peer voice 1 pots destination-pattern 555.... forward-digits extraRelated Commands
destination-pattern
Defines the prefix or the full E.164 telephone number to be used for a dial peer.
show dial-peer voice
Displays configuration information for dial peers.
huntstop
To disable all further dial-peer hunting if a call fails when using hunt groups, enter the huntstop dial-peer configuration command. To reenable dial-peer call hunting, enter the no form of this command.
huntstop
no huntstopSyntax Description
This command has no arguments or keywords.
Defaults
Disabled
Command Modes
Dial-peer configuration
Command History
12.0(5)T
This command was introduced on the Cisco MC3810.
12.0(7)XK
Support for this command was extended to the Cisco 2600 and 3600 series routers.
Usage Guidelines
After you enter this command, no further hunting is allowed if a call fails on the specified dial peer.
This command can be used with all types of dial peers.
Examples
The following example shows how to disable dial-peer hunting on a specific dial peer:
router(config)# dial peer voice 100 vofrrouter(config-dial-peer)# huntstopThe following example shows how to reenable dial-peer hunting on a specific dial peer:
router(config)# dial peer voice 100 vofrrouter(config-dial-peer)# no huntstopRelated Commands
dial-peer voice
Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.
icpif
To specify the Impairment/Calculated Planning Impairment Factor (ICPIF) for calls sent by a dial peer, use the icpif dial peer configuration command. Use the no form of this command to restore the default value for this command.
icpif number
no icpif numberSyntax Description
number
Integer, expressed in equipment impairment factor units, specifying the ICPIF value. Valid entries are from 0 to 55.
Default
The default value for this command is 30.
Command Mode
Dial-peer configuration
Command History
11.3(1)T
This command was introduced on the Cisco 3600 series.
12.0(7)XK
This command was first supported on the Cisco MC3810 platform.
Usage Guidelines
Use the icpif command to specify the maximum acceptable impairment factor for the voice calls sent by the selected dial peer.
This command is applicable only to VoIP peers.
Example
The following example disables the icpif command:
dial-peer voice 10 voipicpif 0incoming called-number
To identify the service type for a call on a router handling both voice and modem calls, use the incoming called-number dial peer configuration command. To return to the default value, use the no form of this command.
incoming called-number string
no incoming called-number stringSyntax Description
string
Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number.
Default
The default value for this command is no associated called number.
Command Mode
Dial peer configuration
Command History
11.3NA
This command was introduced on the Cisco AS5800 platform.
12.0(7)XK
This command was first supported on the Cisco MC3810 platform.
Usage Guidelines
When the Cisco MC3810 is handling both modem and voice calls, it needs to be able to identify the service type of the call—meaning whether the incoming call to the server is a modem or a voice call. When the access server handles only modem calls, the service type identification is handled through modem pools. Modem pools associate calls with modem resources based on the called number (DNIS). In a mixed environment, where the server receives both modem and voice calls, you need to identify the service type of a call by using the incoming called-number command.
If you do not use the incoming called-number command, the server attempts to resolve whether an incoming call is a modem or voice call based on the interface over which the call comes. If the call comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes in over a voice port associated with a dial peer, the call is assumed to be a voice call.
By default, there is no called number associated with the dial peer, which means that incoming calls will be associated with dial peers based on matching calling number with answer address, call number with destination pattern, or calling interface with configured interface.
This command applies to both VoIP and POTS dial peers.
Example
The following example configures calls coming in to the server with a called number of "3799262" as voice calls:
dial peer voice 10 potsincoming called-number 3799262num-exp
To define a complete telephone number for an extension, use the num-exp global configuration command. Use the no form of this command to cancel a configured number expansion.
num-exp extension-number expanded-number
no num-exp extension-numberSyntax Description
extension-number
expanded-number
Digit(s) defining an extension number to be expanded.
Digit(s) defining the expanded telephone number or destination pattern.
Defaults
No number expansion is configured.
Command Mode
Global configuration
Command History
Usage Guidelines
Use the num-exp global configuration command to expand a set of numbers (for example, an extension number) into a destination pattern. With this command, you can map specific extensions and expanded numbers together by explicitly defining each number, or you can define extensions and expanded numbers using variables. You can also use this command to convert seven-digit numbers to numbers containing less than seven digits.
Use a period (.) as a variable or wild card, representing a single number. Use a separate period for each number you want to represent with a wildcard; if you want to replace four numbers in an extension with wildcards, type in four periods.
Example
The following example specifies that extension number 55541 be expanded to 14085555541:
num-exp 55541 14085555541The following example specifies that all five-digit extensions beginning with 5 be expanded to 1408555 . . . .
num-exp 5.... 1408555....Related Commands
session target
To configure a network-specific address for a dial peer, use the session target dial-peer configuration command. Use the no form of this command to disable this feature.
Cisco MC3810 Voice over IP:
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp | loopback:compressed | loopback:uncompressed}no session targetSyntax Description
For the Cisco MC3810 Voice over IP:
Defaults
Enabled with no IP address or domain name defined.
Command Mode
Dial-peer configuration
Command History
Usage Guidelines
This command applies to both the Cisco 3600 series and the Cisco MC3810.
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origination and the loopback type selected.
The session target dns command can be used with or without the specified wildcards. Using the optional wildcards can reduce the number of VoIP dial peer session targets you need to configure if you have groups of numbers associated with a particular router.
Examples
The following example configures a session target using DNS for a host, "voice_router," in the domain "cisco.com":
dial-peer voice 10 voipsession target dns:voice_router.cisco.comThe following example configures a session target using DNS, with the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched portion of the dialed number—in this case, the four-digit extension, to identify the dial peer. As in the previous example, the domain is "cisco.com."
dial-peer voice 10 voipdestination-pattern 1310222....session target dns:$u$.cisco.comThe following example configures a session target using dns, with the optional $d$. wildcard. In this example, the destination pattern has been configured for 13102221111. The optional wildcard $d$. indicates that the router will use the destination pattern to identify the dial peer in the "cisco.com" domain.
dial-peer voice 10 voipdestination-pattern 13102221111session target dns:$d$.cisco.comThe following example configures a session target using DNS, with the optional $e$. wildcard. In this example, the destination pattern has been configured for 12345. The optional wildcard $e$. indicates that the router will reverse the digits in the destination pattern, add periods between the digits, and then use this reverse-exploded destination pattern to identify the dial peer in the "cisco.com" domain.
dial-peer voice 10 voipdestination-pattern 12345session target dns:$e$.cisco.comRelated Commands
show call active voice
To show the active call table, use the show call active voice privileged EXEC command.
show call active voice
Syntax Description
This command has no arguments or keywords.
Command Mode
User EXEC and Privileged EXEC
Command History
Usage Guidelines
This command applies to Voice over IP, Voice over Frame Relay, and Voice over ATM on the Cisco 2600 series, 3600 series, and MC3810 series.
Use this command to display the contents of the active call table, which shows all of the calls currently connected through the router. This command displays information about call times, dial peers, connections, quality of service, and other status and statistical information.
See for a listing of the information types associated with this command.
Example
The following is sample output from the show call active voice command:
router# show call active voiceGENERIC: SetupTime=21072 Index=0 PeerAddress= PeerSubAddress= PeerId=0 PeerIfIndex=0 LogicalIfIndex=0 ConnectTime=0 CallState=3 CallOrigin=2 ChargedUnits=0 InfoType=0 TransmitPackets=375413 TransmitBytes=7508260 ReceivePackets=377734 ReceiveBytes=7554680VOIP: ConnectionId[0x19BDF910 0xAF500007 0x0 0x58ED0] RemoteIPAddress=17635075 RemoteUDPPort=16394 RoundTripDelay=0 SelectedQoS=0 SessionProtocol=1 SessionTarget= OnTimeRvPlayout=0 GapFillWithSilence=0 GapFillWithPrediction=600 GapFillWithInterpolation=0 GapFillWithRedundancy=0 HiWaterPlayoutDelay=110 LoWaterPlayoutDelay=64 ReceiveDelay=94 VADEnable=0 CoderTypeRate=0GENERIC: SetupTime=21072 Index=1 PeerAddress=+14085271001 PeerSubAddress= PeerId=0 PeerIfIndex=0 LogicalIfIndex=5 ConnectTime=21115 CallState=4 CallOrigin=1 ChargedUnits=0 InfoType=1 TransmitPackets=377915 TransmitBytes=7558300 ReceivePackets=375594 ReceiveBytes=7511880TELE: ConnectionId=[0x19BDF910 0xAF500007 0x0 0x58ED0] TxDuration=16640 VoiceTxDuration=16640 FaxTxDuration=0 CoderTypeRate=0 NoiseLevel=0 ACOMLevel=4 OutSignalLevel=-440 InSignalLevel=-440 InfoActivity=2 ERLLevel=227 SessionTarget=provides an alphabetical listing of the fields in this output and a description of each field.
Related Commands
show call history voice
To display the call history table, use the show call history voice privileged EXEC command.
show call history voice [last number | brief]
Syntax Description
Command Mode
User EXEC and Privileged EXEC
Command History
Usage Guidelines
This command applies to all voice applications on the Cisco 2600 series, 3600 series, MC3810, and 7200 series platforms.
Use the show call history voice privileged EXEC command to display the call history table. The call history table contains a listing of all voice calls connected through this router in descending time order. You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword last, and define the number of calls to be displayed with the argument number. To display a shortened version of the call history table, use the keyword brief.
Example
The following is sample output from the show call history voice command for a VoFR call using the frf11-trunk session protocol:
router# show call history voice last 1GENERIC:SetupTime=8283963 msIndex=3149PeerAddress=3623110PeerSubAddress=PeerId=3400PeerIfIndex=18LogicalIfIndex=0DisconnectCause=3FDisconnectText=service or option not available, unspecifiedConnectTime=8283963DisconectTime=8285463CallOrigin=1ChargedUnits=0InfoType=2TransmitPackets=94TransmitBytes=2751ReceivePackets=0ReceiveBytes=0VOFR:ConnectionId=[0x3D4B232D 0x6A900627 0x0 0x4F00852]Subchannel=[Interface Serial0/0, DLCI 160, CID 10]SessionProtocol=frf11-trunkSessionTarget=Serial0/0 160 10CalledNumber=2603100VADEnable=ENABLEDCoderTypeRate=g729r8CodecBytes=30SignalingType=casDTMFRelay=DISABLEDUseVoiceSequenceNumbers=DISABLEDGENERIC:SetupTime=8283963 msIndex=3150PeerAddress=2601100PeerSubAddress=PeerId=1100PeerIfIndex=7LogicalIfIndex=0DisconnectCause=3FDisconnectText=service or option not available, unspecifiedConnectTime=8283964DisconectTime=8285464CallOrigin=2ChargedUnits=0InfoType=2TransmitPackets=0TransmitBytes=-121ReceivePackets=94ReceiveBytes=2563TELE:ConnectionId=[0x3D4B232D 0x6A900627 0x0 0x4F00852]TxDuration=15000 msVoiceTxDuration=2010 msFaxTxDuration=0 msCoderTypeRate=g729r8NoiseLevel=-68ACOMLevel=20SessionTarget=The following is sample output from the show call history voice command for a VoIP call:
router# show call history voiceGENERIC:SetupTime=20405Index=0PeerAddress=PeerSubAddress=PeerId=0PeerIfIndex=0LogicalIfIndex=0DisconnectCause=NORMALDisconnectText=ConnectTime=0DisconectTime=20595CallOrigin=2ChargedUnits=0InfoType=0TransmitPackets=0TransmitBytes=0ReceivePackets=0ReceiveBytes=0VOIP:ConnectionId[0x19BDF910 0xAF500006 0x0 0x56590]RemoteIPAddress=17635075RemoteUDPPort=16392RoundTripDelay=0SelectedQoS=0SessionProtocol=1SessionTarget=OnTimeRvPlayout=0GapFillWithSilence=0GapFillWithPrediction=0GapFillWithInterpolation=0GapFillWithRedundancy=0HiWaterPlayoutDelay=0LoWaterPlayoutDelay=0ReceiveDelay=0VADEnable=0CoderTypeRate=0TELE: ConnectionId=[0x19BDF910 0xAF500006 0x0 0x56590]TxDuration=3030VoiceTxDuration=2700FaxTxDuration=0CoderTypeRate=0NoiseLevel=0ACOMLevel=0SessionTarget=provides an alphabetical listing of the fields in this output and a description of each field.
Related Commands
show num-exp
To show the number expansions configured, use the show num-exp privileged EXEC command.
show num-exp [dialed-number]
Syntax Description
Command Mode
User EXEC and Privileged EXEC
Command History
Usage Guidelines
This command applies to VoFR, VoATM, and Voice over IP on the Cisco 2600 series, 3600 series, and MC3810 platforms.
Use the show num-exp privileged EXEC command to display all of the number expansions configured for this router. To display number expansion for only one number, specify that number by using the dialed-number argument.
Example
The following is sample output from the show num-exp command:
router# show num-expDest Digit Pattern = '0...' Translation = '+14085270...'Dest Digit Pattern = '1...' Translation = '+14085271...'Dest Digit Pattern = '3..' Translation = '+140852703..'Dest Digit Pattern = '4..' Translation = '+140852804..'Dest Digit Pattern = '5..' Translation = '+140852805..'Dest Digit Pattern = '6....' Translation = '+1408526....'Dest Digit Pattern = '7....' Translation = '+1408527....'Dest Digit Pattern = '8...' Translation = '+14085288...'explains the fields in the sample output.
Related Commands
voice class codec
To enter voice-class configuration mode and assign an identification tag number for a codec voice class, use the voice class codec global configuration command. Use the no form of this command to delete a codec voice class.
voice class codec tag
no voice class codec tagSyntax Description
tag
The unique number you assign to the voice class. The valid range is 1 to 10000. Each tag number must be unique on the router.
Command Modes
Global configuration
Command History
Usage Guidelines
This command only creates the voice class for codec selection preference, and assigns an identification tag. Use the codec preference command to specify the parameters of the voice class, and use the voice-class codec dial-peer command to apply the voice class to a VoIP dial peer.
Note
Example
The following example shows how to enter voice-class configuration mode and assign a voice class tag number starting from global configuration mode:
router(config)# voice class codec 10router(config-class)#After you enter voice-class configuration mode for codecs, use the codec preference command to specify the parameters of the voice class.
Related Commands
Defines the order of preference in which network dial peers select codecs.
Assigns a previously-configured codec selection preference list to a dial peer.
voice-class codec (dial-peer)
To assign a previously-configured codec selection preference list (codec voice class) to a VoIP dial peer, enter the voice-class codec dial-peer configuration command. Enter the no form of this command to remove the codec preference assignment from the dial peer.
voice-class codec tag
no voice-class codec tagSyntax Description
tag
The unique number assigned to the voice class. The valid range for this tag is 1 to 10000. The tag number maps to the tag number created using the voice class codec global configuration command.
Defaults
Dial peers have no codec voice class assigned.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
You can assign one voice class to each VoIP dial peer. If you assign another voice class to a dial peer, the last voice class assigned replaces the previous voice class.
Note
Examples
The following example shows how to assign a previously-configured codec voice class to a dial peer:
router(config)# dial-peer voice 100 voiprouter(config-dial-peer)# voice-class codec 10Related Commands
voice-group
This command was added in Cisco IOS Release 11.3(1)MA on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
