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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.
Voice over IP offers the following benefits:
This feature supports the following standards and RFCs:
The voice enhancements described in this document require the use of Cisco IOS Release 12.0(7)XK or newer.
To configure Voice over IP on the Cisco MC3810 concentrator, you need to complete the following tasks:
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:
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.
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:
(a) POTS—Dial peer describing the characteristics of a traditional telephony network connection. POTS peers point to a particular voice port on a voice network device. To minimally configure a POTS dial peer, you need to configure the following two characteristics: associated telephone number and logical interface. Use the destination-pattern command to associate a telephone number with a POTS peer. Use the port command to associate a specific logical interface with a POTS peer. In addition, you can specify direct inward dialing for a POTS peer by using the direct-inward-dial command.
(b) VoIP—Dial peer describing the characteristics of a packet network connection; in the case of Voice over IP, this is an IP network. VoIP peers point to specific VoIP devices. To minimally configure a VoIP peer, you need to configure the following two characteristics: associated destination telephone number and a destination IP address. Use the destination-pattern command to define the destination telephone number associated with a VoIP peer. Use the session target command to specify a destination IP address for a VoIP peer.
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:
(a) FXO—Foreign Exchange Office interface
(b) FXS—The Foreign Exchange Station interface
(c) E&M—The "Ear and Mouth" interface (or "RecEive and TransMit" interface)
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.
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.
Before you can configure your Cisco MC3810 concentrator to use Voice over IP, you must first:
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.
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:
In general, backbone routers perform the following QoS functions:
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:
Each of these components is discussed in the following sections.
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:
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:
To configure Multilink PPP and interleaving on a configured and operational interface or virtual interface template, use the following commands in interface mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| | Enable Multilink PPP. | ||
| | Enable real-time packet interleaving. | ||
| | Optionally, configure a maximum fragment delay. | ||
| | Reserve a strict priority queue for a set of RTP packet flows belonging to a range of UDP destination ports |
For more information about Multilink PPP, refer to the "Configuring Media-Independent PPP and Multilink PPP" chapter in the Dial Solutions Configuration Guide.
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 1 ppp multilink encapsulated ppp ppp multilink interleave ppp multilink fragment-delay 20 ip rtp priority 16384 16383 25 multilink virtual-template 1
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 Figure 1.
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).
You should configure RTP header compression if the following conditions exist in your network:
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:
| Command | Purpose |
|---|---|
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.
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:
| Command | Purpose |
|---|---|
Specify the total number of RTP header compression connections supported on an interface. |
The following example enables RTP header compression for a serial interface:
interface 0 ip rtp header-compression encapsulation ppp ip rtp compression-connections 25
For more information about RTP header compression, see the "Configuring IP Multicast Routing" chapter of the Network Protocols Configuration Guide, Part 1.
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.
| Command | Purpose |
|---|---|
router(config-if)# ip rtp priority starting-rtp-port-number port-number-range bandwidth | Reserves a strict priority queue for a set of RTP packet flows belonging to a range of UDP destination ports. |
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.
| Command | Purpose |
|---|---|
| (Optional) If using the number expansion feature, define a destination pattern for an extension number. Repeat for each extension to be expanded. |
This section describes how to use new commands defining dial-peer operation for Voice over IP on Cisco MC3810 series concentrators.
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:
| Command | Purpose |
|---|---|
| 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:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| | Define the telephone number associated with this POTS dial peer. Note | ||
| | Associate this POTS dial peer with a specific voice port. |
To configure direct inward dial (DID) for a particular POTS dial peer, use the following commands beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| | Enter dial-peer configuration mode to configure a POTS peer. | ||
| | Specify direct inward dial for this POTS peer. |
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.
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:
| Command | Purpose |
|---|---|
| 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:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| | Define the destination telephone number associated with this VoIP dial peer. | ||
| | Specify a destination IP address for this dial peer. | ||
| router(config-dialpeer)# dtmf-relay [cisco-rtp] [h245-signal] [h245-alphanumeric] | (Optional) 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. Note This command is only supported if your Cisco MC3810 has version 549 or newer DSPs. |
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."
You can check the validity of your dial-peer configuration by performing the following tasks:
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.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router(config)# dial-peer hunt | (Optional) Specify the hunting selection order for dial peers. | ||
| | (Optional) Designate a terminating character for variable length dialed numbers. The default character is # (pound sign). |
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:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router(config)# dial-peer voice tag {pots | voip} | Enter dial-peer configuration mode for the specified dial peer. | ||
| router(config-dial-peer)# huntstop | Disable dial-peer hunting on the dial peer. Once you enter this command, no further hunting will be allowed if a call fails on the specified dial peer. |
To reenable dial-peer hunting on a dial peer, enter the no huntstop command.
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.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| or or | (Optional) If using the forward-digits feature, configure the digit-forwarding method. The range for the number of digits forwarded (num-digit) is 0 to 32. Refer to the command reference section for an explanation of the command options. In the default condition, dialed digits not matching the destination pattern are forwarded. Note The no state is not the default state. | ||
| | (Optional) If the forward-digits feature was not configured in the last step, assign the dialed digits prefix for the dial peer. | ||
| | (Optional) Configure a preference for the POTS dial peer. The value is a number from 0 (highest preference) to 10 (lowest preference). If POTS and voice-network (VoFR, VoATM, VoIP) dial peers are mixed in the same hunt group, POTS dial peers will be searched first, even if a voice-network peer has a higher preference number. |
To give real-time voice traffic precedence over other IP network traffic, use the following commands, beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| | Enter the dial-peer configuration mode to configure a VoIP peer. | ||
| | Select a precedence level for the voice traffic associated with that dial peer. |
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 voip ip precedence 5
In 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.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| | Enter the dial-peer configuration mode to configure a VoIP peer. | ||
| | Specify the desired voice coder rate of speech.Optionally specify the voice payload (in bytes) of each frame. |
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 voip destination-pattern +14085551234 codec g711alaw session target ipv4:10.0.0.8
| Step | Command | Purpose | ||
|---|---|---|---|---|
| | Enter dial-peer configuration mode to configure a VoIP peer. | ||
| | Disable the transmission of silence packets (enabling VAD). |
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 voip destination-pattern +14085551234 vad session target ipv4:10.0.0.8
To configure codec selection order, perform the following tasks:
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:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router(config)# voice class codec tag | Create a voice class for a codec preference list. The range for the tag number is 1 to 10000. The tag number must be unique on the router. | ||
| router(config-voice-class)# codec preference priority codec [bytes payload-size] | Configure the selection order of preference for a codec. Repeat this command to specify selection orders of preference for additional codecs, if required. | ||
| router(config-voice-class) #exit | Exit from voice-class configuration mode. |
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:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router(config)# dial-peer voice tag voip |
The tag is a number that identifies the dial peer and must be unique on the router. Do not assign duplicate tag numbers. | ||
|
| Assign to the dial peer the voice class that you created in the "Configuring a Voice Class to Define Codec Selection Order" section. Note The voice-class command in dial-peer configuration mode is entered with a hyphen. The voice class command in global configuration mode is entered without the hyphen. |
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-config Building configuration... Current configuration: ! version 12.0 . . . voice class codec 10 codec preference 1 g711alaw codec preference 2 g711ulaw bytes 80 codec preference 3 g726r16 bytes 120 ! voice class codec 20 codec preference 1 g726r24 bytes 90 codec preference 2 g726r32 bytes 120 ! voice class codec 30 codec preference 1 g729ar8 codec preference 2 g726r16 codec preference 3 g726r32 ! . . . dial-peer voice 101 voip voice-class codec 10 ! dial-peer voice 102 voip voice-class codec 20 ! dial-peer voice 103 voip voice-class codec 30 ! line con 0 transport input none line aux 0 line 2 3 line vty 0 4 password #1writer login ! end
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:
Under most circumstances the default values are adequate for FXO and FXS voice ports.
If you need to change the default configuration for these voice ports, perform the following tasks:
1. Configure the applicable parameters for the voice port.
2. Verify the configuration.
3. Troubleshoot and correct any configuration errors.
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.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router(config)# voice-port slot/port | Identify the voice port you want to configure and enter voice-port configuration mode. | ||
| router(config-voice-port)#connection {plar | tie-line | trunk | plar-opx} string | Specify the voice-port connection type and the destination telephone number.
| ||
| router(config-voice-port)#voice confirmation-tone | If connection plar or connection plar-opx is configured, enable the two-beep confirmation tone that a caller hears when picking up the handset. | ||
| router(config-voice-port)#dial-type {dtmf | pulse} | (FXO only) Select the dial type for dialing out.
| ||
| router(config-voice-port)#signal {loop-start | ground-start} | (Analog only) Select the appropriate signaling type. | ||
| router(config-voice-port)#cptone country | Select the appropriate call progress tone for your country location. The default is northamerica. For a list of supported countries, refer to the Voice, Video, and Home Applications Command Reference. | ||
| router(config-voice-port)#compand-type {u-law | a-law} | Configure the companding standard used to convert between analog and digital signals in PCM systems. Defaults are: u-law for T1; a-law for E1. | ||
| router(config-voice-port)#vad | (Optional) Enable voice activity detection (VAD). | ||
| router(config-voice-port)#comfort-noise | (Optional) Enable background noise if VAD is enabled. | ||
| router(config-voice-port)#music-threshold number | (Optional) Specify the maximum volume (in dBm) for on-hold music. Valid entries are -70 to -30. | ||
| router(config-voice-port)#description string | (Optional) Describe the location, connected equipment, or other information about the voice port. The description is displayed when a show command is entered. | ||
| router(config-voice-port)#exit | Exit from voice-port configuration mode. | ||
| router(config)# voice-card 0 | Enter voice-card configuration mode and specify voice card 0. Voice card 0 provides the configuration mode for setting the codec complexity on a Cisco MC3810. | ||
| router(config-voicecard)# codec complexity {high | medium}
| Specify the codec complexity for this Cisco MC3810 according to the bandwidth requirements and the number of voice channels to be supported per DSP. The default is medium complexity, which provides four voice channels per DSP. Note You cannot change codec complexity while DS0 groups are defined. If they are already set up, use the no ds0-group command before resetting the codec complexity. | ||
| router(config-voice-ca)#exit | Exit from voice-card configuration mode. | ||
| router(config-voice-port)#exit | Exit from voice-port configuration mode. |
You can check the validity of your voice-port configuration by performing the following tasks:
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:
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.
To fine tune FXO and FXS voice ports, perform the following tasks:
1. Perform the voice-port tuning procedure for the voice port.
2. Verify the configuration.
3. Troubleshoot and correct any configuration errors.
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.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router(config)#voice-port slot/port | Identify the voice port you want to configure and enter voice-port configuration mode. | ||
| router(config-voiceport)#input gain value | Specify the receive gain (in dB) for the voice port. Value range is -6 to 14. | ||
| router(config-voiceport)#output attenuation value | Specify the transmit attenuation (in dB) for the voice port. Value range is 0 to 14. | ||
| router(config-voiceport)#echo-cancel enable | Enable echo-cancellation of voice that is sent out the interface and received back on the same interface. | ||
| router(config-voiceport)#echo-cancel coverage {16 | 24 | 32} | Set the duration (in milliseconds) of echo cancellation. Values are 16, 24, and 32. | ||
| router(config-voiceport)#non-linear | Enable non-linear processing, which shuts off any signal if no near-end speech is detected. (Non-linear processing is used with echo-cancellation.) | ||
| router(config-voiceport)#playout-delay | Tune the playout buffer to accommodate packet jitter caused by switches in the WAN. | ||
| | (For T1/E1 digital voice ports only.) Configure the voice port to manipulate the transmit and/or receive bit patterns to match the bit patterns required by a connected device. Be careful not to destroy the information content of the bit pattern. For example, forcing the A-bit on or off will prevent FXO interfaces from being able to generate both an on-hook and off-hook state. Note The show voice port command reports at the protocol level, while the show controller command reports at the driver level. The driver is not notified of any bit manipulation using the condition command. As a result, the show controller command output will not account for the bit conditioning. | ||
| router(config-voiceport)# timeouts initial seconds | Specify the number of seconds the system waits for a caller to dial the first digit. The range is 10 to 120. The default is 10. | ||
| router(config-voiceport)# timeouts interdigit seconds | Specify the number of seconds the system waits, after a caller has dialed the initial digit, for the caller to dial each subsequent digit. The range is 0 to 120. The default is 10. | ||
| router(config-voiceport)# timeouts ringing {seconds | infinity} | Specify the maximum number of seconds that a voice port allows ringing to continue if a call is not answered. The range is 5 to 60000. The default is 180. | ||
| router(config-voiceport)# timeouts wait-release {seconds | infinity} | Specify the maximum number of seconds that a voice port can remain in the call failure state while the router or concentrator sends a busy tone, reorder tone, or out-of-service tone to the port. The value range is 5 to 3600. The default is 30. | ||
| router(config-voiceport)# timing digit milliseconds
| If the dial type is DTMF, configure the DTMF digit signal duration in milliseconds. The range is 50 to 100. The default is 100. | ||
| router(config-voiceport)# timing inter-digit milliseconds | |||
| router(config-voiceport)# timing pulse-digit milliseconds | |||
| router(config-voiceport)# timing pulse-inter-digit milliseconds
| |||
| router(config-voiceport)# timing percentbreak percent | (FXO only) Specify the percentage of the break period for dialing pulses. The range is 20 to 80. The default is 50. | ||
| router(config-voiceport)# timing guard-out milliseconds | (FXO only) Specify the duration in milliseconds of the guard-out period to prevent this port from seizing a remote FXS port before the remote port detects a disconnect signal. The range is 300 to 3000. The default is 2000. | ||
| router(config-voiceport)# impedance {600r | 600c | 900r | 900c} | (FXO only) Configure the impedance. The default is 600r (600 ohms real). | ||
| router(config-voiceport)# ring number number | (Analog FXO only) Configure the number of rings detected before a call is answered on the FXO port. The range is 1 to 10. The default is 1. | ||
| router(config-voiceport)# ring frequency number | (FXS only) Specify the local ring frequency (Hertz) for the FXS voice port. Valid entries are 20 and 30. The default is 20. | ||
| router(config-voiceport)# disconnect-ack | (FXS only) Configure the voice port to return an acknowledgment upon receipt of a disconnect signal. | ||
| router(config-voiceport)# ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07 | pattern08 | pattern09 | pattern10 | pattern11 | pattern12 ] [define pulse-interval]} | (FXS only) Specify the on and off times for the ringing pulses. See the command reference section for details on the ring cadence options. | ||
| router(config-voiceport)#exit | Exit from voice-port configuration mode. |
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.
To configure E&M voice ports, perform the following tasks:
1. Configure the applicable parameters for the voice port.
2. Verify the configuration.
3. Troubleshoot and correct any configuration errors.
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.
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router(config)# voice-port slot/port | Identify the voice port you want to configure and enter voice-port configuration mode. | ||
| router(config-voiceport)# connection {plar | tie-line | trunk | plar-opx} destination-string [answer-mode] | Specify the voice-port connection type and the destination telephone number.
When configuring Cisco-trunk permanent calls, one side must be the call initiator (master) and the other side is normally the call answerer (slave). By default, the voice port operates in master mode. Enter the answer-mode keyword to specify that the voice port should operate in slave mode. | ||
| router(config-voiceport)# voice confirmation-tone | If connection plar-opx is configured, enable the two-beep confirmation tone that a caller hears when picking up the handset. | ||
| router(config-voiceport)# dial-type {dtmf | pulse | mf } | Select the dial type for dialing out.
| ||
| router(config-voiceport)# operation {2-wire | 4-wire} | Select the appropriate cabling scheme for this voice port. | ||
| router(config-voiceport)# type {1 | 2 | 3 | 5} | Select the appropriate E&M interface type. Type 1 lead configuration:
Type 2 lead configuration:
Type 3 lead configuration:
Type 5 lead configuration:
| ||
| router(config-voiceport)# signal {wink-start | immediate | delay-dial} | Configure the E&M signaling type. The default is wink-start. | ||
| router(config-voiceport)# cptone country | Select the appropriate call progress tone for your country location. The default is northamerica. For a list of supported countries, refer to the Voice, Video, and Home Applications Command Reference. | ||
| router(config-voiceport)# compand-type {u-law | a-law} | Configure the companding standard used to convert between analog and digital signals in PCM systems. Defaults are: u-law for T1; a-law for E1. | ||
| router(config-voiceport)# no vad | (Optional) Disable voice activity detection (VAD). VAD is enabled by default. | ||
| router(config-voiceport)# comfort-noise | (Optional) Enable background noise if VAD is enabled. | ||
| router(config-voiceport)# music-threshold number | (Optional) Specify the maximum volume (in dBm) for on-hold music. Valid entries are -70 to -30. The default is -38. | ||
| router(config-voiceport)# voice confirmation-tone | (Optional) If the voice port is configured for connection plar-opx for Off-Premises eXtension, disable the two-beep confirmation tone that a caller hears when picking up the handset. | ||
| router(config-voiceport)# description string | (Optional) Describe the location, connected equipment, or other information about the voice port. The description is displayed when a show command is entered. | ||
| router(config-voice-port)#exit | Exit from voice-port configuration mode. |
You can check the validity of your voice-port configuration by performing the following tasks:
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:
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.
To fine tune E&M voice ports, perform the following tasks:
1. Perform the voice-port tuning procedure for the voice port.
2. Verify the configuration.
3. Troubleshoot and correct any configuration errors.
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.
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:
| Command | Purpose |
|---|---|
router(config-voiceport)# no shutdown | Activate the voice port. |
To cycle a voice port, enter the following commands in voice-port configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router(config-voiceport)# shutdown | Deactivate the voice port. | ||
| router(config-voiceport)# voice-port slot/port | Identify the voice port you want to activate and enter the voice-port configuration mode. | ||
| router(config-voiceport)# no shutdown | Activate the voice port. | ||
| router(config-voice-port)#exit | Exit from voice-port configuration mode. |
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
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.
Enable VoIP gateway functionality by using the gateway command.
To enable gateway functionality, use the following commands:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router# configure terminal | Enter global configuration mode. | ||
| router(config)# gateway | Enable the VoIP gateway. |
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:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| router(config)# interface type slot/port | Enter interface configuration mode to configure parameters for the specified interface. | ||
| router(config-if)# ip address ip-address subnet-mask | Specify the IP address for this interface. | ||
| router(config-if)#h323-gateway voip interface | Designate this interface as the H.323 gateway interface. | ||
| router(config-if)#h323-gateway voip h323-id interface-id | Specify an H.323 name (ID) for the gateway associated with this interface. This ID is used by this gateway when this gateway communicates with the gatekeeper. Usually, this H.323 ID is the name given to the gateway with the gatekeeper domain name appended to the end. | ||
| router(config-if)#h323-gateway voip id gatekeeper {ipaddr ip-address [port]| multicast} | Specify the name (ID) of the gatekeeper associated with this gateway and how the gateway finds it. The gatekeeper ID configured here must exactly match the gatekeeper ID in the gatekeeper configuration. The gateway determines the location of the gateway in one of two ways: either by a defined IP address or through multicast. | ||
| router(config-if)#h323-gateway voip tech-prefix prefix | Specify a technology prefix. A technology prefix is used to identify a type of service that this gateway is capable of providing. Note If a gateway is capable of handling multiple services, specify each service with a tech-prefix command. | ||
| router(config-if)#exit | Exit interface configuration mode. | ||
| router(config)#exit | Exit global configuration mode. |
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:
These examples are described in the following sections. The following examples use the term "router" to generically describe Cisco routers and concentrators.
The following example shows how to configure Voice over IP to link PBX users with E&M trunk lines.
Figure 2 illustrates the topology of this connection example.
hostname sanjose !Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern 555.... port 1/0/0 !Configure pots dial peer 2 dial-peer voice 2 pots destination-pattern 555.... port 1/0/1 !Configure voip dial peer 3 dial-peer voice 3 voip destination-pattern 119.... session target ipv4:172.16.65.182 !Configure the E&M interface voice-port 1/0/0 signal immediate operation 4-wire type 2 voice-port 1/0/1 signal immediate operation 4-wire type 2 !Configure the serial interface interface serial 0/0 description serial interface type dce (provides clock) clock rate 2000000 ip address 172.16.1.123 no shutdown
hostname saltlake !Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern 119.... port 1/0/0 !Configure pots dial peer 2 dial-peer voice 2 pots destination-pattern 119.... port 1/0/1 !Configure voip dial peer 3 dial-peer voice 3 voip destination-pattern 555.... session target ipv4:172.16.1.123 !Configure the E&M interface voice-port 1/0/0 signal immediate operation 4-wire type 2 voice-port 1/0/0 signal immediate operation 4-wire type 2 !Configure the serial interface interface serial 0/0 description serial interface type dte ip address 172.16.65.182 no shutdown
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.
Figure 3 illustrates the topology of this connection example.
! Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern +14085554000 port 1/0/0 ! Configure voip dial peer 2 dial-peer voice 2 voip destination-pattern 9........... session target ipv4:172.16.65.182 ! Configure the serial interface interface serial 0/0 clock rate 2000000 ip address 172.16.1.123 no shutdown
! Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern 9........... port 1/0/0 ! Configure voip dial peer 2 dial-peer voice 2 voip destination-pattern +14085554000 session target ipv4:172.16.1.123 ! Configure serial interface interface serial 0/0 ip address 172.16.65.182 no shutdown
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.
Figure 4 illustrates the topology of this connection example.
! Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern +14085554000 port 1/0/0 ! Configure voip dial peer 2 dial-peer voice 2 voip destination-pattern 9........... session target ipv4:172.16.65.182 ! Configure the serial interface interface serial 0/0 clock rate 2000000 ip address 172.16.1.123 no shutdown
! Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern 9........... port 1/0/0 ! Configure voip dial peer 2 dial-peer voice 2 voip destination-pattern +14085554000 session target ipv4:172.16.1.123 ! Configure the voice-port voice-port 1/0/0 connection plar 14085554000 ! Configure the serial interface interface serial 0/0 ip address 172.16.65.182 no shutdown
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 10 router(config-class)# codec preference 1 g711alaw router(config-class)# codec preference 2 g711ulaw bytes 80 router(config-class)# codec preference 3 g723ar53 router(config-class)# codec preference 4 g723ar63 bytes 144 router(config-class)# codec preference 5 g723r53 router(config-class)# codec preference 6 g723r63 bytes 120 router(config-class)# codec preference 7 g726r16 router(config-class)# codec preference 8 g726r24 router(config-class)# codec preference 9 g726r32 bytes 80 router(config-class)# codec preference 10 g728 router(config-class)# codec preference 11 g729br8 router(config-class)# codec preference 12 g729r8 bytes 50 router(config-class)# exitrouter(config-class)#exit router(config)#
The following example assigns a voice class 10 to a VoIP dial peer:
router(config)# dial-peer voice 25 voip router(config-dial-peer)# voice-class codec 10
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.
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
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.
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. |
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.
Voice class configuration
| Release | Modification |
|---|---|
12.0(2)XH | This command was introduced on the Cisco AS5300. |
12.0(7)T | This command was first supported on the Cisco 2600 and 3600 series routers. |
12.0(7)XK | This command was first supported on the Cisco MC3810 series. |
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.
| Codec | Protocol | Voice Payload Options (bytes) | Default Voice Payload (bytes) |
|---|---|---|---|
| g711alaw g711ulaw | VoIP | 80, 160 | 160 |
| g723ar53 g723r53 | VoIP | 20 to 220 in multiples of 20 | 20 |
| g723ar63 g723r63 | VoIP | 24 to 216 in multiples of 24 | 24 |
| g726r16 | VoIP | 20 to 220 in multiples of 20 | 40 |
| g726r24 | VoIP | 30 to 210 in multiples of 30 | 60 |
| g726r32 | VoIP | 40 to 200 in multiples of 40 | 80 |
g728 | VoIP | 10 to 230 in multiples of 10 | 40 |
g729abr8 | VoIP | 10 to 230 in multiples of 10 | 20 |
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 10 router(config-class)# codec preference 1 g711alaw router(config-class)# codec preference 2 g711ulaw bytes 80 router(config-class)# codec preference 3 g723ar53 router(config-class)# codec preference 4 g723ar63 bytes 144 router(config-class)# codec preference 5 g723r53 router(config-class)# codec preference 6 g723r63 bytes 120 router(config-class)# codec preference 7 g726r16 router(config-class)# codec preference 8 g726r24 router(config-class)# codec preference 9 g726r32 bytes 80 router(config-class)# codec preference 10 g728 router(config-class)# codec preference 11 g729br8 router(config-class)# codec preference 12 g729r8 bytes 50 router(config-class)# exitrouter(config)#exit router)#
| Command | Description |
Enters voice-class configuration mode and assigns an identification tag number for a codec voice class. | |
Assigns a previously-configured codec selection preference list to a dial peer. |
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]}
plar | Specifies a private line auto ring down (PLAR) connection. PLAR is handled by associating a peer directly with an interface; when an interface goes off-hook, the peer is used to set up the second call leg and conference them together without the caller having to dial any digits. |
tie-line | Specifies a tie-line connection to a private branch exchange (PBX). |
plar-opx | Specifies a PLAR Off-Premises eXtension connection. Using this option, the local voice-port provides a local response before the remote voice-port receives an answer. On FXO interfaces, the voice-port will not answer until the remote side answers. |
digits | The destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number. |
trunk | Specifies a straight tie-line connection to a private branch exchange (PBX). |
answer-mode | (Optional; used only with the trunk keyword.) Specifies that the router should not attempt to initiate a trunk connection, but should wait for an incoming call before establishing the trunk. |
No connection mode is specified.
Voice-port configuration
| Release | Modification |
|---|---|
11.3(1)T | This command was first introduced. |
11.3(1)MA1 | This command was first supported on the Cisco MC3810, and the tie-line keyword was first made available on the Cisco MC3810. |
11.3(1)MA5 and 12.0(2)T | The plar-opx keyword was first made available on the Cisco MC3810 as the plar-opx-ringrelay keyword. The keyword was shortened in a subsequent release. |
12.0(3)XG and 12.0(4)T | The trunk keyword was made available on the Cisco MC3810. The trunk answer-mode option was added. |
12.0(7)XK | This command options were unified across the Cisco 2600, 3600, and MC3810 platforms. |
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.
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.
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 5559262
The 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 5559262
The 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 5559262
The 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-mode
| Command | Description |
destination-pattern | Specifies either the prefix or the full E.164 telephone number to be used for a dial peer. |
dial-peer voice | Enters dial-peer configuration mode and specifies the method of voice-related encapsulation. |
session-protocol | Establishes a session protocol for calls between the local and remote routers via the packet network. |
session-target | Configures a network-specific address for a dial peer. |
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
hunt-order-number | A number from 0 to 7 that selects a predefined hunting selection order: 0—Longest match in phone number, explicit preference, random selection. This is the default hunt order number. 1—Longest match in phone number, explicit preference, least recent use. 2—Explicit preference, longest match in phone number, random selection. 3—Explicit preference, longest match in phone number, least recent use. 4—Least recent use, longest match in phone number, explicit preference. 5—Least recent use, explicit preference, longest match in phone number. 6—Random selection. 7—Least recent use. |
The default is longest match in phone number, explicit preference, random selection (hunt order number 0).
Global configuration
| Release | Modification |
|---|---|
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. |
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.
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 terminal dial-peer hunt 0
| Command | Description |
destination-pattern | Specifies the prefix or the complete telephone number for a dial peer. |
preference | Specifies the preferred selection order of a dial peer within a hunt group. |
show dial-peer voice | Displays configuration information for dial peers. |
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
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 #. |
The default terminating character is #.
Global configuration
| Release | Modification |
|---|---|
12.0 | This command was introduced. |
12.0(7)XK | Usage was restricted to variable-length dialed numbers. |
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.
The following example specifies "9" as the terminating character for variable-length dialed numbers:
configure terminal dial-peer terminator 9#
| Command | Description |
answer-address | Specifies the preferred selection order of a dial peer within a hunt group. |
destination-pattern | Specifies the prefix or the complete telephone number for a dial peer. |
timeouts interdigit | Specifies the interdigit timeout value for a voice port, in seconds. |
show dial-peer voice | Displays configuration information for dial peers. |
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}For the Cisco 3600 series:
dial-peer voice tag {pots | voip | voatm | vofr }For the Cisco MC3810 series:
dial-peer voice tag {pots | voip | voatm | vofr }
tag | A number identifying a particular dial peer. Valid entries are 1 to 2147483647. |
pots | POTS dial peer using basic telephone service. |
voip | VoIP dial peer using voice encapsulation on the POTS network. |
voatm | (Cisco 3600 and MC3810 only) Voice over ATM dial peer using real-time AAL5 voice encapsulation on the ATM backbone network. |
vofr | Voice over Frame Relay dial peer using encapsulation on the Frame Relay backbone network. |
No default behavior or values.
Global configuration
| Release | Modification |
|---|---|
11.3(1)T | This command was first introduced. |
11.3(1)MA | This command was first supported on the Cisco MC3810, with support for POTS, VoFR, and VoATM. |
12.0(3)XG and 12.0(4)T | This command added VoFR to the Cisco 2600 and 3600 series routers. |
12.0(4)T | This command added VoFR to the Cisco 7200 series platform. |
12.0(7)XK | This command added VoIP to the Cisco MC3810 and VoATM to the Cisco 3600 series routers. |
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.
The following example accesses dial-peer configuration mode and configures a POTS peer identified as dial peer 10:
configure terminal dial-peer voice 10 pots
| Command | Description |
voice-port | Enters voice-port configuration mode. |
ds0-group ds0-group-no timeslots timeslot-list type signal-type
no ds0-group ds0-group-no
ds0-group-no | A value from 0 to 23 that identifies the DS0 group. |
timeslot-list | timeslot-list is a single timeslot number, a single range of numbers, or multiple ranges of numbers separated by commas. For T1, allowable values are from 1 to 24. Examples are: · 2 · 1-15, 17-24 · 1-23 · 2, 4, 6-12 |
type | The signaling method selection for type depends on the connection that you are making. The E&M interface allows connection for PBX trunk lines (tie- lines) and telephone equipment. The FXS interface allows connection of basic telephone equipment and PBXs. The FXO interface is for connecting the central office (CO) to a standard PBX interface where permitted by local regulations. The FXO interface is often used for off-premises extensions. The options are as follows: · e&m-immediate-start—no specific off-hook and on-hook signaling · e&m-delay-dial—the originating endpoint sends an off-hook signal and then waits for an off-hook signal followed by an on-hook signal from the destination · e&m-wink-start—the originating endpoint sends an off-hook signal and waits for a wink signal from the destination · fxs-ground-start—Foreign Exchange Station ground-start signaling support · fxs-loop-start —Foreign Exchange Station loop-start signaling support · fxo-ground-start—Foreign Exchange Office ground-start signaling support · fxo-loop-start—Foreign Exchange Office loop-start signaling support |
| The following options are available only on E1 controllers on the Cisco MC3810: · e&m-melcas-immed—E&M Mercury Exchange Limited Channel Associated Signaling (MELCAS) immediate start signaling support · e&m-melcas-wink—E&M MELCAS wink start signaling support · e&m-melcas-delay—E&M MELCAS delay start signaling support · fxo-melcas—MELCAS Foreign Exchange Office signaling support · fxs-melcas—MELCAS Foreign Exchange Station signaling support |
| The following options are available only when the mode ccs command is enabled on the Cisco MC3810 for transparent CCS support: · ext-sig-master—For the specified channel(s), automatically generates the off-hook signal and stays in the off-hook state. · ext-sig-slave—For the specified channel(s), automatically generates the answer signal when a call is terminated to that channel. |
No DS0 group is defined.
Controller configuration
| Release | Modification |
|---|---|
11.2 | This command was introduced for the Cisco AS5300 as cas-group. |
12.0(1)T | The cas-group command was first supported on the Cisco 3600 series. |
12.0(5)T | This command was renamed ds0-group on the Cisco AS5300 and on the Cisco 2600 and 3600 series (requires Digital T1 Packet Voice Trunk Network Modules). |
12.0(7)XK | Support for this command was extended to the Cisco MC3810. When the ds0-group command became available on the Cisco MC3810, the voice-group command was removed and is no longer supported. |
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.
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 1/0 router(config-controller)# framing esf router(config-controller)# linecode b8zs router(config-controller)# ds0-group 1 timeslot 1-10 type fxs-ground-start router(config-controller)# ds0-group 2 timeslot 11-24 type fxo-loop-start
The 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-slave
| Command | Description |
codec complexity | Matches the DSP complexity packaging to the codec(s) to be supported |
mode ccs | Configures the T1/E1 controller to support CCS cross-connect or CCS frame-forwarding. |
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]
cisco-rtp | (Optional) Forwards DTMF tones using RTP protocol with a Cisco proprietary payload type. |
h245-signal | (Optional) Forwards DTMF tones using the H.245 "signal" User Input Indication method. Supports tones 0-9, *, #, and A-D. |
h245-alphanumeric | (Optional) Forwards DTMF tones using the H.245 "alphanumeric" User Input Indication method. Supports tones 0-9, *, #, and A-D. |
DTMF tones are sent "inband", or left in the audio stream, unless you use this command.
EXEC
| Release | Modification |
|---|---|
11.3(2) NA | This command was introduced. |
12.0(5)T | This command was modified for H.323 V2, adding dtmf-relay and h245-signal. |
12.0(7)XK | This command is supported on the Cisco MC3810 |
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. cisco-rtp (highest priority)
2. h245-signal
3. h245-alphanumeric
4. None - DTMF sent inband
The following are two examples of the dtmf-relay command:
Router# configure terminal Router(config)# dial-peer voice 103 voip Router(config-dial-peer)# dtmf-relay cisco-rtp h245-signal Router(config-dial-peer)# end Router#
Router# configure terminal Router(config)# dial-peer voice 103 voip Router(config-dial-peer)# no dtmf-relay Router(config-dial-peer)# end
| Command | Description |
dial-peer | Switch to the voice-port configuration mode form the global configuration mode. |
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}
num-digit | The number of digits to be forwarded. If the number of digits is greater than the length of a destination phone number, the length of the destination number is used. The valid range is 0 to 32. Setting the value to 0 is equivalent to entering no forward-digits. |
all | Forward all digits. If all is entered, the full length of the destination pattern is used. |
extra | If the length of the dialed digit string is greater than the length of the dial-peer destination pattern, the extra right-justified digits are forwarded. However, if the dial-peer destination pattern is variable length (ending with character T, for example: T, 123T, 123...T), extra digits are not forwarded. |
Dialed digits not matching the destination-pattern are forwarded.
Dial-peer configuration
| Release | Modification |
|---|---|
11.3(1) MA | This command was first introduced on the Cisco MC3810. |
12.0(2) T | The implicit option was added. |
12.0(4) T | This command was modified to support ISDN PRI QSIG signaling calls. |
12.0(7)XK | This command was first supported on the Cisco 2600 series and 3600 series platforms, the implicit keyword was removed, and the extra keyword was added. |
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.
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 all
The 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 4
The 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 extra
| Command | Description |
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. |
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.
huntstopThis command has no arguments or keywords.
Disabled
Dial-peer configuration
| Release | Modification |
|---|---|
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. |
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.
The following example shows how to disable dial-peer hunting on a specific dial peer:
router(config)# dial peer voice 100 vofr
router(config-dial-peer)# huntstop
The following example shows how to reenable dial-peer hunting on a specific dial peer:
router(config)# dial peer voice 100 vofr
router(config-dial-peer)# no huntstop
| Command | Description |
dial-peer voice | Enters dial-peer configuration mode and specifies the method of voice-related encapsulation. |
number | Integer, expressed in equipment impairment factor units, specifying the ICPIF value. Valid entries are from 0 to 55. |
The default value for this command is 30.
Dial-peer configuration
| Release | Modification |
|---|---|
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. |
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.
The following example disables the icpif command:
dial-peer voice 10 voip icpif 0
string | Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number. |
The default value for this command is no associated called number.
Dial peer configuration
| Release | Modification |
|---|---|
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. |
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.
The following example configures calls coming in to the server with a called number of "3799262" as voice calls:
dial peer voice 10 pots incoming called-number 3799262
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
extension-number expanded-number | Digit(s) defining an extension number to be expanded. Digit(s) defining the expanded telephone number or destination pattern. |
No number expansion is configured.
Global configuration
| Release | Modification |
|---|---|
11.3(1)T | This command was first introduced on the Cisco 3600 platform. |
12.0(3)T | This command was first supported on the Cisco AS5300 platform. |
12.0(4)XL | This command was first supported on the Cisco AS5800 platform. |
12.0(7)XK | This command was first supported on the Cisco MC3810 platform. |
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.
The following example specifies that extension number 55541 be expanded to 14085555541:
num-exp 55541 14085555541
The following example specifies that all five-digit extensions beginning with 5 be expanded to 1408555 . . . .
num-exp 5.... 1408555....
| Command | Description |
forward-digits | Specifies which digits to forward for voice calls. |
prefix | Specifies a prefix for a dial peer. |
dial-peer terminator | Change the character used as a terminator for variable length dialed numbers. |
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 target
For the Cisco MC3810 Voice over IP:
ipv4:destination-address | IP address of the dial peer. |
dns:host-name | Indicates that the domain name server will be used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device. (Optional) You can use one of the following three wildcards with this keyword when defining the session target for VoIP peers:
|
loopback:rtp | Indicates that all voice data will be looped back to the originating source. This is applicable for VoIP peers. |
loopback:compressed | Indicates that all voice data will be looped back in compressed mode to the originating source. This is applicable for POTS peers. |
loopback:uncompressed | Indicates that all voice data will be looped-back in uncompressed mode to the originating source. This is applicable for POTS peers. |
Enabled with no IP address or domain name defined.
Dial-peer configuration
| Release | Modification |
|---|---|
11.3(1) T | This command was first introduced. |
11.3(1) MA | Support was added for VoFR,VoATM and VoHDLC dial peers on the Cisco MC38110. |
12.0(3) XG and 12.0(4)T | The cid option was added. Support was added for VoFR dial peers on the Cisco 2600 and Cisco 3600 series routers. |
12.0(7)XK | Support was added for VoATM dial peers on the Cisco 3600 series routers. Support was added for VoIP dial peers on the Cisco MC3810. Support for VoHDLC on the Cisco MC3810 was removed in this release. |
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.
The following example configures a session target using DNS for a host, "voice_router," in the domain "cisco.com":
dial-peer voice 10 voip session target dns:voice_router.cisco.com
The 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 voip destination-pattern 1310222.... session target dns:$u$.cisco.com
The 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 voip destination-pattern 13102221111 session target dns:$d$.cisco.com
The 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 voip destination-pattern 12345 session target dns:$e$.cisco.com
| Command | Description |
called-number | Enables an incoming VoFR call leg to be bridged to the correct POTS call leg. |
codec (dial-peer) | Specifies the voice coder rate of speech for a dial peer. |
cptone | Specifies a regional tone, ring, and cadence setting for an analog voice port. |
destination-pattern | Specifies either the prefix or the full E.164 telephone number to be used for a dial peer. |
dtmf-relay | Enables the DSP to generate FRF.11 Annex A frames for a dial peer. |
preference | Indicates the preferred selection order of a dial peer within a hunt group. |
session protocol | Establishes a VoFR protocol for calls between the local and the remote routers via the packet network. |
To show the active call table, use the show call active voice privileged EXEC command.
show call active voiceThis command has no arguments or keywords.
User EXEC and Privileged EXEC
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced on the Cisco 2600 series and 3600 series. |
12.0(3)XG | Support for VoFR was added. |
12.0(4)T | This command was first supported on the Cisco 7200 series. |
12.0(7)XK | This command was first supported on the Cisco MC3810 platform. |
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 Table 2 for a listing of the information types associated with this command.
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=7554680
VOIP: 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=0
GENERIC: 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=7511880
TELE: 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=
Table 2 provides an alphabetical listing of the fields in this output and a description of each field.
| Field | Description |
|---|---|
ACOM Level | Current ACOM level for the call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call. |
CallOrigin | Call origin; answer versus originate. |
CallState | Current state of the call. |
CoderTypeRate | Negotiated coder transmit rate of voice/fax compression during the call. |
ConnectionId | Global call identifier of a gateway call. |
ConnectTime | Time at which the call was connected. |
Dial-Peer | Tag of the dial peer transmitting this call. |
ERLLevel | Current Echo Return Loss (ERL) level for this call. |
FaxTxDuration | Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value. |
GapFillWithSilence | Duration of voice signal replaced with silence because voice data was lost or not received on time for this call. |
GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call. An example of such pullout is frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms. |
GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received on time from voice gateway for this call. |
GapFillWithRedundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because voice data was lost or not received on time from voice gateway for this call. |
HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during this call. |
Index | Dial peer identification number. |
InfoActivity | Active information transfer activity state for this call. |
InfoType | Information type for this call. |
InSignalLevel | Active input signal level from the telephony interface used by this call. |
LogicalIfIndex | Index number of the logical interface for this call. |
LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during the call. |
NoiseLevel | Active noise level for the call. |
OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
OutSignalLevel | Active output signal level to telephony interface used by this call. |
PeerAddress | Destination pattern associated with this peer. |
PeerId | ID value of the peer table entry to which this call was made. |
PeerIfIndex | Voice port index number for this peer. |
PeerSubaddress | Subaddress to which this call is connected. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the decoder delay during the voice call. |
ReceivePackets | Number of packets received by this peer during this call. |
RemoteIPAddress | Remote system IP address for the VoIP call. |
RemoteUDPPort | Remote system UDP listener port to which voice packets are transmitted. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone during the call. |
SelectedQoS | Selected quality of service (QoS) for the call. |
SessionProtocol | Session protocol used for an Internet call between the local and remote router via the IP backbone. |
SessionTarget | Session target of the peer used for the call. |
SetupTime | Value of the System UpTime when the call associated with this entry was started. |
TransmitBytes | Number of bytes transmitted from this peer during the call. |
TransmitPackets | Number of packets transmitted from this peer during the call. |
TxDuration | Duration of transmit path open from this peer to the voice gateway for the call. |
VADEnable | Whether or not voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice transmission from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value. |
| Command | Description |
Displays the call history table. | |
show dial-peer voice | Displays configuration information for dial peers. |
Displays the number expansions configured. | |
show voice port | Displays configuration information about a specific voice port. |
To display the call history table, use the show call history voice privileged EXEC command.
show call history voice [last number | brief]
last number | (Optional) Displays the last calls connected, where the number of calls displayed is defined by the argument number. Valid entries for the argument number are numbers from 1 to 2147483647. |
brief | (Optional) Displays abbreviated call history information for each leg of a call. |
User EXEC and Privileged EXEC
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced on the Cisco 3600 series. |
12.0(3)XG | Support for VoFR was added on the Cisco 2600 and 3600 series. |
12.0(4)T | The brief keyword was added and the command was first supported on the Cisco 7200 series. |
12.0(7)XK | Support for the brief keyword was added on the Cisco MC3810 platform. |
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.
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 ms Index=3149 PeerAddress=3623110 PeerSubAddress= PeerId=3400 PeerIfIndex=18 LogicalIfIndex=0 DisconnectCause=3F DisconnectText=service or option not available, unspecified ConnectTime=8283963 DisconectTime=8285463 CallOrigin=1 ChargedUnits=0 InfoType=2 TransmitPackets=94 TransmitBytes=2751 ReceivePackets=0 ReceiveBytes=0 VOFR: ConnectionId=[0x3D4B232D 0x6A900627 0x0 0x4F00852] Subchannel=[Interface Serial0/0, DLCI 160, CID 10] SessionProtocol=frf11-trunk SessionTarget=Serial0/0 160 10 CalledNumber=2603100 VADEnable=ENABLED CoderTypeRate=g729r8 CodecBytes=30 SignalingType=cas DTMFRelay=DISABLED UseVoiceSequenceNumbers=DISABLED GENERIC: SetupTime=8283963 ms Index=3150 PeerAddress=2601100 PeerSubAddress= PeerId=1100 PeerIfIndex=7 LogicalIfIndex=0 DisconnectCause=3F DisconnectText=service or option not available, unspecified ConnectTime=8283964 DisconectTime=8285464 CallOrigin=2 ChargedUnits=0 InfoType=2 TransmitPackets=0 TransmitBytes=-121 ReceivePackets=94 ReceiveBytes=2563 TELE: ConnectionId=[0x3D4B232D 0x6A900627 0x0 0x4F00852] TxDuration=15000 ms VoiceTxDuration=2010 ms FaxTxDuration=0 ms CoderTypeRate=g729r8 NoiseLevel=-68 ACOMLevel=20 SessionTarget=
The following is sample output from the show call history voice command for a VoIP call:
router#show call history voiceGENERIC: SetupTime=20405 Index=0 PeerAddress= PeerSubAddress= PeerId=0 PeerIfIndex=0 LogicalIfIndex=0 DisconnectCause=NORMAL DisconnectText= ConnectTime=0 DisconectTime=20595 CallOrigin=2 ChargedUnits=0 InfoType=0 TransmitPackets=0 TransmitBytes=0 ReceivePackets=0 ReceiveBytes=0 VOIP: ConnectionId[0x19BDF910 0xAF500006 0x0 0x56590] RemoteIPAddress=17635075 RemoteUDPPort=16392 RoundTripDelay=0 SelectedQoS=0 SessionProtocol=1 SessionTarget= OnTimeRvPlayout=0 GapFillWithSilence=0 GapFillWithPrediction=0 GapFillWithInterpolation=0 GapFillWithRedundancy=0 HiWaterPlayoutDelay=0 LoWaterPlayoutDelay=0 ReceiveDelay=0 VADEnable=0 CoderTypeRate=0 TELE: ConnectionId=[0x19BDF910 0xAF500006 0x0 0x56590] TxDuration=3030 VoiceTxDuration=2700 FaxTxDuration=0 CoderTypeRate=0 NoiseLevel=0 ACOMLevel=0 SessionTarget=
Table 3 provides an alphabetical listing of the fields in this output and a description of each field.
| Field | Description |
|---|---|
ACOMLevel | Average ACOM level for this call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call. |
CallOrigin | Call origin; answer versus originate. |
CoderTypeRate | Negotiated coder rate. This value specifies the transmit rate of voice/fax compression to its associated call leg for the call. |
ConnectionID | Global call identifier for the gateway call. |
ConnectTime | Time the call was connected. |
DisconnectCause | Description explaining why the call was disconnected. |
DisconnectText | Descriptive text explaining the disconnect reason. |
DisconnectTime | Time the call was disconnected. |
FaxDuration | Duration of fax transmitted from this peer to the voice gateway for this call. You can derive the Fax Utilization Rate by dividing this value by the TxDuration value. |
GapFillWithSilence | Duration of voice signal replaced with silence because the voice data was lost or not received on time for this call. |
GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call. |
GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call. |
GapFillWithRedundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because the voice data was lost or not received on time from the voice gateway for this call. |
HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during the voice call. |
Index | Index number identifying the voice-peer for this call. |
InfoType | Information type for this call. |
LogicalIfIndex | Index of the logical voice port for this call. |
LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during the voice call. |
NoiseLevel | Average noise level for this call. |
OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
PeerAddress | Destination pattern or number to which this call is connected. |
PeerId | ID value of the peer entry table to which this call was made. |
PeerIfIndex | Index number of the logical interface through which this call was made. For ISDN media, this would be the index number of the B channel used for the call. |
PeerSubAddress | Subaddress to which this call is connected. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the decoder delay during the voice call. |
ReceivePackets | Number of packets received by this peer during the call. |
RemoteIPAddress | Remote system IP address for the call. |
RemoteUDPPort | Remote system UDP listener port to which voice packets for this call are transmitted. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone for this call. |
SelectedQoS | Selected quality of service for the call. |
SessionProtocol | Session protocol to be used for an Internet call between the local and remote router via the IP backbone. |
SessionTarget | Session target of the peer used for the call. |
SetUpTime | Value of the System UpTime when the call associated with this entry was started. |
TransmitBytes | Number of bytes transmitted by this peer during the call. |
TransmitPackets | Number of packets transmitted by this peer during the call. |
TxDuration | Duration of the transmit path open from this peer to the voice gateway for the call. |
VADEnable | Whether or not voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice transmitted from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration by the TxDuration value. |
| Command | Description |
Displays the contents of the active call table. | |
show dial-peer voice | Displays configuration information for dial peers. |
Displays the number expansions configured. | |
show voice port | Displays configuration information about a specific voice port. |
To show the number expansions configured, use the show num-exp privileged EXEC command.
show num-exp [dialed-number]
dialed-number | (Optional) Dialed number. |
User EXEC and Privileged EXEC
| Release | Modification |
|---|---|
11.3(1)T | This command was first introduced on the Cisco 3600 platform. |
12.0(3)T | This command was first supported on the Cisco AS5300 platform. |
12.0(4)XL | This command was first supported on the Cisco AS5800 platform. |
12.0(7)XK | This command was first supported on the Cisco MC3810 platform. |
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.
The following is sample output from the show num-exp command:
router# show num-exp Dest 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...'
Table 4 explains the fields in the sample output.
| Field | Description |
|---|---|
Dest Digit Pattern | Index number identifying the destination telephone number digit pattern. |
Translation | Expanded destination telephone number digit pattern. |
| Command | Description |
Displays the contents of the active call table. | |
Displays the call history table. | |
show dial-peer voice | Displays configuration information for dial peers. |
show voice port | Displays configuration information about a specific voice port. |
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
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. |
Global configuration
| Release | Modification |
|---|---|
12.0(2)XH | This command was introduced on the Cisco AS5300. |
12.0(7)T | This command was first supported on the Cisco 2600 and 3600 series routers. |
12.0(7)XK | This command was first supported on the Cisco MC3810 series. |
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.
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 10
router(config-class)#
After you enter voice-class configuration mode for codecs, use the codec preference command to specify the parameters of the voice class.
| Command | Description |
Defines the order of preference in which network dial peers select codecs. | |
Assigns a previously-configured codec selection preference list to a 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
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. |
Dial peers have no codec voice class assigned.
Dial-peer configuration
| Release | Modification |
|---|---|
12.0(2)XH | This command was introduced on the Cisco AS5300. |
12.0(7)T | This command was first supported on the Cisco 2600 and 3600 series routers. |
12.0(7)XK | This command was first supported on the Cisco MC3810 series. |
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.
The following example shows how to assign a previously-configured codec voice class to a dial peer:
router(config)# dial-peer voice 100 voip router(config-dial-peer)# voice-class codec 10
| Command | Description |
Defines the order of preference in which network dial peers select codecs. | |
Enters voice-class configuration mode and assigns an identification tag number for a codec voice class. | |
show dial-peer voice | Displays the configuration for all dial peers configured on the router. |
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.
Posted: Wed Sep 27 16:33:39 PDT 2000
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