Table Of Contents
Land Mobile Radio over IP Enhancement
Prerequisites for Land Mobile Radio over IP Enhancement
Restrictions for Land Mobile Radio over IP Enhancement
Information About Land Mobile Radio over IP Enhancement
LMR Feature Enhancements in Cisco IOS Release 12.4(15)XY
Improved Debugging Capabilities
LED Troubleshooting Enhancement
Use of E-Lead and M-Lead Signaling
How to Configure Land Mobile Radio over IP Enhancement
Configuring a Digital LMR Voice Port
Configuring an Analog LMR Voice Port
Configuring Polarity and Additional Restrictions on the LMR Voice Port
Configuring Polarity and Additional Restrictions on the M-Lead
Configuring Polarity and Additional Restrictions on the E-Lead
Configuring Connections Between LMR Routers
Adjusting the Voice Quality on the LMR Voice Port
Verifying Land Mobile Radio over IP Enhancement
Configuration Examples for Land Mobile Radio over IP Enhancement
Configuring Connection Trunk on an Analog LMR Voice Port: Example
Verifying Connection Trunk on an Analog LMR Voice Port: Example
Configuring Connection Trunk on a Digital LMR Voice Port: Example
Verifying Connection Trunk on a Digital LMR Voice Port: Example
Configuring PLAR on an Analog LMR Voice Port: Example
Verifying PLAR on an Analog LMR Voice Port: Example
Configuring PLAR on a Digital LMR Voice Port: Example
Verifying PLAR on a Digital LMR Voice Port: Example
Configuring VoIPmc with Connection Trunk on an Analog LMR Voice Port: Example
Verifying VoIPmc with Connection Trunk on an Analog LMR Voice Port: Example
Configuring VoIPmc with Connection Trunk on a Digital LMR Voice Port: Example
Verifying VoIPmc with Connection Trunk on a Digital LMR Voice Port: Example
Configuring VoIPmc with Connection PLAR on an Analog LMR Voice Port: Example
Verifying VoIPmc with Connection PLAR on an Analog LMR Voice Port: Example
Configuring VoIPmc with Connection PLAR on a Digital LMR Voice Port: Example
Verifying VoIPmc with Connection PLAR on a Digital LMR Voice Port: Example
Land Mobile Radio over IP Enhancement
The Land Mobile Radio over IP Enhancement feature allows Cisco multiservice routers to transport Land Mobile Radio (LMR) traffic over IP networks by modifying voice gateway functionality. LMR over IP enables LMR systems to extend beyond their traditional geographic limitations created by transmitter signal strength and enables interoperability, allowing public safety personnel in different agencies or jurisdictions to communicate with each other by radio on demand, in real time.
Note
Some support restrictions apply to use of the Cisco Land Mobile Radio (LMR) over IP feature. See the "DISCLAIMER" section for important information regarding Cisco support for this feature.
Throughout this document, references to LMR radios apply to all types of radios, including LMR, military, amateur, and others.Feature History for Land Mobile Radio over IP Enhancement
Finding Support Information for Platforms and Cisco IOS Software Images
Use Cisco Feature Navigator to find information about platform support and Cisco IOS software image support. Access Cisco Feature Navigator at http://www.cisco.com/go/fn. You must have an account on Cisco.com. If you do not have an account or have forgotten your username or password, click Cancel at the login dialog box and follow the instructions that appear.
Contents
•
Prerequisites for Land Mobile Radio over IP Enhancement
•
Restrictions for Land Mobile Radio over IP Enhancement
•
Information About Land Mobile Radio over IP Enhancement
•
How to Configure Land Mobile Radio over IP Enhancement
•
Configuration Examples for Land Mobile Radio over IP Enhancement
Prerequisites for Land Mobile Radio over IP Enhancement
•
Install the appropriate Cisco IOS images on each router. Table 1 lists the images that support the Land Mobile Radio over IP Enhancement feature. Land Mobile Radio over IP Enhancement features require the spservicesk9 image at a minimum.
•
Make sure you have the required amount of memory on each router. Table 2 lists the memory requirements for various platform images.
•
Configure a working VoIP network.
Restrictions for Land Mobile Radio over IP Enhancement
•
VIC-2E/M voice interface cards (VICs) work with the NM-2V network modules only.
•
VIC2-2E/M VICs work with the NM-HD-2V and NM-HD-2VE network modules only.
•
The NM-2Vnetwork module is not supported on the Cisco 2800 series and Cisco 3800 series platforms.
•
When performing the following configuration tasks remember that even after you issue the correct configuration commands, some configurations require a new call to be set up or a shutdown command then a no shutdown command to be issued before taking effect. Table 3 shows the configuration items, their associated commands, and the events that must occur for some of the configuration items to take effect. In Table 3, the values in the New Call Set Up and Shut, No Shut columns have the following meanings:
–
A Yes in the New Call Set Up column indicates that the configuration task does not take effect until a new call is set up.
–
A No in the New Call Set Up column indicates that the configuration task takes effect without a new call being set up.
–
A Yes in the Shut, No Shut column indicates that the configuration task does not take effect until a shtudown command then a no shutdown command is issued on the port.
–
A No in the Shut, No Shut column indicates that the configuration task takes effect without a shtudown command then a no shutdown command being issued on the port.
Information About Land Mobile Radio over IP Enhancement
To configure Land Mobile Radio over IP Enhancement, you need to understand the following concepts:
•
LMR Feature Enhancements in Cisco IOS Release 12.4(15)XY
•
Use of E-Lead and M-Lead Signaling
LMR Feature Enhancements in Cisco IOS Release 12.4(15)XY
This section describes the new features in Cisco IOS Release 12.4(15)XY.
Bootup Without Radio Keying
In this release, the router can boot and reboot without keying the attached radio. This capability is present only with a specific combination of VIC and Cisco IOS software release. The behavior of the various VIC and Cisco IOS software release combinations is described in Table 4.
Transmit Delay
LMR gateways are prone to front-end clipping when they are connecting to a trunked radio system because of the time required to acquire a channel. This feature provides a configurable delay before the voice packet is played out to compensate for the channel acquisition time. The maximum delay is 1.5 seconds. The transmit delay is available for LMR ports only.
Tone Injection
Many conventional radio systems use in-band tone signaling to indicate activity, key the transmitter, and control channel selection. There are three phases of tone signaling:
•
Wakeup tone—A tone of a specific duration and frequency that acts as preamble to base stations to indicate that additional signaling is coming.
•
Frequency selection (or control) tone—One of a range of tones used to select a frequency (channel) for the audio.
•
Guard tone— A tone of a specific frequency that is maintained as long as there is activity on the channel. This tone indicates that the channel has been seized.
To eliminate the need for tones to be passed across the WAN, this feature provides the capability to inject tones at the gateway. Static tone injection is one fixed sequence of single tones, no more than ten tones or pauses in a given sequence, used on all transmissions from that voice port to the attached LMR system. Static tone injection begins with E-lead activity and ends when the hangover time expires on voice playout. The tone sequence comprises some combination of the following:
•
Single tone—Of fixed frequency, duration, and amplitude.
•
Pause—Of fixed duration.
•
Guard tone—Of fixed frequency and amplitude. To be played out with the voice packet, for the duration of the voice packet.
•
Idle tone—To be played in the absence of voice packets. Idle tone and guard tone are mutually exclusive.
If you configure injected tones, be sure to use the command to configure a delay before the voice packet is played out. Configuring a delay prevents the voice packet from being overwritten by the injected tones. The delay must be equal to the sum of the durations of the injected tones and pauses in the tone-signal voice class.
Digital Filter
The digital filter improves voice quality by preventing transmission of the guard tone from the LMR system to the VoIP network. The digital filter can be configured to filter out either 2175 Hz or 1950 Hz through the command. Only one of these frequencies can be filtered out at a time. Filtering is performed by the digital signal processor (DSP). Digital filtering is disabled by default. The digital filter is available for LMR ports only.
Improved Debugging Capabilities
The command enhances debugging capability by providing LMR-related dynamic and static information along with detailed voice port and active call information, eliminating the need to use several different debugging commands. Information displayed by the show voice lmr command allows for improved troubleshooting of the interface between the LMR gateways and LMR radio systems.
LED Troubleshooting Enhancement
The default behavior of the E&M LED is to indicate voice activity only. You can now configure the LED to provide industry standard behavior using the command. The LED indicates transmission and reception as follows:
•
Red—Voice from the network toward the radio is active (E-lead)
•
Green—Voice from the radio towards the network is active (M-lead or voice activity detection (VAD))
•
Yellow—Voice in both directions is active
Configurable PTT Timeout
To limit extended radio transmission, a configurable Push To Talk (PTT) timeout feature has been added. The PTT timeout can be configured for different durations, up to 30 minutes, on a port-by-port basis using the command. This timeout can be configured on LMR ports only.
Automatic Gain Control
Because of radio network loss and other environmental factors, the speech level arriving at a router from an LMR system can be very low. Automatic gain control, which is performed by the DSP, adjusts speech to a comfortable volume when it becomes too loud or too soft. You can use automatic gain control to ensure that the speech is played back at a more comfortable level. Because the gain is inserted digitally, the background noise can also be amplified. You can configure input gain and output attenuation in decibels per milliwatt (dBm). Automatic gain control can be configured on LMR ports only and is mutually exclusive with set input-gain function.
Connection Types
The Land Mobile Radio over IP Enhancement feature works with the connection types discussed in the following sections:
•
PLAR
Connection Trunk
LMR features can be integrated into traditional point-to-point trunk connections. VoIP simulates two types of trunk connections—switched and permanent—that can be configured for both analog and digital systems. Switched connections are discussed in the "PLAR" section. The connections are created with the connection command. Refer to the Cisco IOS Voice Command Reference, Release 12.3 T, for a description of the connection command.
The connection trunk command creates a permanent call that is connected as soon as the E&M voice ports on the routers on each end are brought up (see Figure 1). Permanent calls pass limited telephony signaling and operate without collecting digits or requiring changes to the overall dial plan.
Figure 1 Connection Trunk Configuration
The calls simulate a permanent tie-line between a radio system and its dispatch console. Both ends must be configured for E&M voice port signaling.
PLAR
Private line, automatic ringdown (PLAR) is a switched simulated connection that can be configured for both analog and digital systems. LMR features can be integrated into the traditional point-to-point PLAR connections. When a switched call is configured (see Figure 2), the user can make a call without dialing any digits. The router uses the digits configured with the connection plar command internally to send the call to a dial peer.
Figure 2 Connection PLAR Configuration
The switched call configuration works with any type of voice port (ear and mouth (E&M), Foreign Exchange Office (FXO), or Foreign Exchange Station (FXS)) and can be used without any effect on an existing dial plan. Switched call configuration is commonly used to connect PBXs in which the remote devices appear to be physical extensions.
VoIP Multicast
VoIP multicast (VoIPmc) networks provide "always on" multiuser conferences without requiring that users dial in to a conference. By using the inherent point-to-multipoint connectivity of IP multicast (IPmc), the routers can take several inbound voice streams and forward the packetized voice over the IP network to all parties within a defined VoIPmc group. In LMR systems, VoIPmc can connect more than two radios and is required if an IP-based dispatcher application is used to mix and manage different radio channels.
Cisco's VoIP technology, which was initially focused on traditional PBX toll-bypass applications, can be used to combine VoIPmc networks with data networks. VoIP's characteristic dynamic sharing of bandwidth is even more compelling with VoIPmc than with a toll-bypass application because in an LMR environment the relatively short, infrequent bursts of voice activity leave ample bandwidth available for data applications during the long periods of inactivity.
Figure 3 shows a diagram of the Cisco VoIPmc solution connecting legacy equipment over an IP network.
Note
The "V" on the Cisco router icons signifies that some of the VoIPmc bridging function is being done by the router's digital signal processors (DSPs).
Figure 3 VoIPmc Using Cisco 3725 and Cisco 3745 Routers
Use of E-Lead and M-Lead Signaling
The Land Mobile Radio over IP Enhancement feature allows you to define the use of the E-lead and M-lead in signaling between the E&M voice port on the router and the attached LMR device. The E-lead connects to the Push To Talk (PTT) of the LMR system as shown in Figure 4. The M-lead corresponds to the Carrier Operated Relay (COR) of the LMR system, which indicates receive activity on the LMR system. You can change how the E-lead and M-lead signals are used to suit the needs of your LMR system with the and commands.
Figure 4 E-Lead and M-Lead Connections
The lmr e-lead command has the following options:
•
Inactive—The router never sends a seize signal on the E-lead to the LMR device. The router sends voice packets to LMR devices. Use this option if you are connecting a tone-controlled radio to the router.
•
Seize—The router sends a seize signal on the E-lead when the LMR port is connected and removes the seize signal from the E-lead when the LMR port is not involved in a VoIP connection. This is the default. Use this option if your radio requires PTT operation.
•
Voice—The router sends a seize signal on the E-lead only when it receives voice packets from the network. When no packets are detected on the network, the seize signal is removed from the E-lead. This option is the same as voice operated transmit (VOX).
The command has the following options:
•
Inactive—The router ignores signals sent by voice on the M-lead. The flow of voice packets is determined by VAD. The router sends voice received from the LMR device. This is the default. This option is the same as tone control or VOX.
•
Audio-gate-in—The router generates VoIP packets when a seize signal is detected on the M-lead. The router stops generating VoIP packets when the seize signal is removed from the M-lead. An LMR voice port configured for audio-gate-in cannot initiate a PLAR connection.
•
Dialin—When the LMR device is not involved in a VoIP connection, the first seize signal detected on the M-lead triggers the router to set up a VoIP connection. This behavior gives the ability for activity on the radio COR to trigger a VoIP call to another VoIP endpoint. Once the connection is made, the router behaves as described in the audio-gate-in option, which is the same as tone control or VOX. The VoIP connection can then remain active indefinitely, or it can time out because of inactivity based on the timer set with the command. Use this option with PLAR connections only.
Polarity
The Land Mobile Radio over IP Enhancement feature allows you to configure the voice port to match E&M bit patterns with the attached LMR device. E&M interfaces use two-state signaling, in which the interface is in either seize or idle state for both transmit and receive. E-lead signal polarity is independent from M-lead signal polarity. In normal polarity, the idle bit pattern is 0000, and the seize bit pattern is 1111. In reverse polarity, the idle bit pattern is 1111, and the seize bit pattern is 0000.
An LMR device with PTT functionality usually looks for an open relay contact, which is normal polarity. However, some LMR devices look for a closed relay pattern, which is reverse polarity.
You can customize the seize and idle patterns with the define command. For analog voice ports, bit patterns are not usually customized. Customizing the bit patterns for reverse polarity is a special circumstance reserved for LMR signaling.
In LMR systems that use connection trunk connections, the M-lead signal is sent to the far-end router as a keepalive signal, which the far-end router plays out. If you do not want the M-lead signal played out, define the seize bit pattern to be the same as the idle bit pattern to make sure that only the idle signal will be sent to the far-end router and that the M-lead signal is ignored.
Virtual Interface
In all Cisco VoIPmc implementations, the routers are configured with an "interface vif1." This is a virtual interface that is similar to a loopback interface—a logical IP interface that is always up when the router is active. In addition, it must be configured so the Cisco VoIPmc packets that are locally mixed on the DSPs can be fast-switched along with the other data packets. This interface must reside on its own unique subnet, using a 30-bit subnet mask. The virtual interface uses two of the subnet addresses. The virtual interface subnet should be included in the routing protocol updates (Routing Information Protocol [RIP], Open Shortest Path First [OSPF], and so on). The virtual interface should not be used as the source address for protocols. Loopback addresses should be used instead.
In this example of the interface vif command, the resulting multicast source address is the address above the interface address. In this example, the VoIPmc source would be 192.168.5.2.
interface Vif1ip address 192.168.5.1 255.255.2555.252ip pim sparse-dense-modeE&M Signaling Types
Note
This section describes only E&M signaling Type II, Type III, and Type V. Cisco routers do not support Type IV, and Type I is not conducive to LMR.
Type II is preferred for use with LMR because the absence of DC connectivity between radios and the router ensures that no ground loops are created.Cisco LMR routers support E&M signaling Type II, Type III, and Type V. With each signaling type, the router supplies one signal, known as the M signal (for Mouth), and accepts one signal, known as the E signal (for Ear). Conversely, the LMR equipment accepts the M signal from the router and provides the E signal to the router. The M signal accepted by the LMR equipment at one end of a circuit becomes the E signal output by the remote LMR interface.
Figures 5 through 7 show the interface models for the different E&M signaling types supported for LMR. Table 5 explains terms used in the figures.
Figure 5 E&M Type II Interface Model
The interface model shown in Figure 5 is correct for a dry relay contact closure for COR functionality. For open collector or open drain outputs, you would have to wire through a user-supplied applique before connecting to the LMR over IP router.
CautionFailure to add an applique may result in damage to radio equipment by the -48 VDC present on the SB lead.
Figure 6 E&M Type III Interface Model
Figure 7 E&M Type V Interface Model
Note
Using Type V will create a ground loop between router and radio, which may impact performance.
Codecs
Cisco VoIP gateways use coder-decoders (codecs), which are DSP software algorithms used to compress and decompress speech or audio signals.
Some codec compression techniques require more processing power than others. Codec complexity is broken into two categories, medium and high complexity. The difference between medium and high complexity codecs is the amount of CPU utilization necessary to process the codec algorithm, and therefore, the number of voice channels that can be supported by a single DSP. Medium complexity codecs support four channels per DSP. High complexity codecs support two channels per DSP. For this reason, all the medium complexity codecs can also be run in high complexity mode, but fewer (usually half) of the channels are available per DSP.
Refer to Understanding Codecs: Complexity, Hardware Support, MOS, and Negotiation for a list of network modules and the codecs they support.
VAD Tuning
Cisco voice activity detection (VAD) has two layers: application programming interface (API) layer and processing layer. There are three states into which the processing layer classifies incoming signals: speech, unknown, and silence. The state of the incoming signals is determined by the noise threshold, which can be configured with the threshold noise command.
If the voice level is below the noise threshold, then the signal is classified as silence. If the incoming signal cannot be classified, the variable thresholds that are computed with the statistics of speech and noise that VAD gathers are used to make a determination. If the signal still cannot be classified, then it is marked as unknown. The final decision is made by the API. In some applications, you could have the noise create unwanted spurious packets (for example, a voice stream) taking up bandwidth.
Speech and unknown packets are sent over the network; silence packets are discarded. The sound quality of the connection is slightly degraded with VAD, but the connection monopolizes much less bandwidth. If VAD is disabled, voice data is continuously sent to the IP backbone.
When the aggressive keyword is used with the vad command in dial peer configuration mode, the VAD noise threshold is reduced from -78 to -62 dBm. Noise that falls below the -62 dBm threshold is considered to be silence and is not sent over the network. Additionally, unknown packets are considered to be silence and are discarded.
The music-threshold command specifies the decibel level of music played when calls are put on hold. This command tells the firmware to pass steady data above the specified level. The music threshold affects only the operation of VAD when the voice port is receiving voice.
For more information on VAD tuning, refer to Troubleshooting Hissing and Static.
How to Configure Land Mobile Radio over IP Enhancement
This section contains the following procedures:
•
Configuring an LMR Voice Port (required)
•
Configuring Polarity and Additional Restrictions on the LMR Voice Port (optional)
•
Configuring Tone Signaling (optional)
•
Configuring Connections Between LMR Routers (required)
•
Adjusting the Voice Quality on the LMR Voice Port (optional)
•
Verifying Land Mobile Radio over IP Enhancement (optional)
Configuring an LMR Voice Port
An LMR voice port is similar to an E&M voice port with Immediate Start signaling and auto-cut-through enabled. Perform one of the following tasks to create an LMR voice port:
•
Configuring a Digital LMR Voice Port (optional)
•
Configuring an Analog LMR Voice Port (optional)
LMR Basics
When the LMR system sends voice to the LMR router, Cisco IOS software detects either that the LMR port M-lead is on or that the VAD status has changed. When an LMR voice port receives voice, either the Cisco IOS software turns on the LMR voice port E-lead or a third-party application sends tone on the voice path.
An LMR voice port on a PLAR connection cannot initiate a call unless the dialin option of the command was used to configure dial-in capability. LMR PLAR connections are torn down manually with the test lmr clear-call command or upon expiration of a teardown timer set with the command.
Duplex Mode
You can configure the LMR voice port to operate in half-duplex or full-duplex mode. Configuring half-duplex mode helps avoid noise being fed back into the network. The duplex mode is on a per port basis. Cisco IOS software does not prevent full-duplex ports from talking to half-duplex ports.
Configuring a Digital LMR Voice Port
Perform this task to configure a digital LMR voice port.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
controller t1 slot/port
4.
ds0-group ds0-group-number timeslots timeslot-list type e&m-lmr
5.
exit
6.
voice-port slot/port:ds0-group-number
7.
shutdown
8.
lmr duplex half
9.
lmr led-on
10.
timing ignore m-lead milliseconds
11.
timing delay-voice tdm milliseconds
12.
timeout ptt {rcv | xmt} minutes
13.
no comfort-noise
14.
no echo-cancel enable
15.
no shutdown
16.
end
DETAILED STEPS
Configuring an Analog LMR Voice Port
Perform this task to configure an analog LMR voice port.
Prerequisites
For analog LMR voice ports, the voice interface card must be an E&M card.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port slot-number/subunit-number/port
4.
shutdown
5.
type {1 | 2 | 3 | 5}
6.
operation {2-wire | 4-wire}
7.
signal lmr
8.
lmr duplex half
9.
lmr led-on
10.
timing ignore m-lead milliseconds
11.
timing delay-voice tdm milliseconds
12.
timeout ptt {rcv | xmt} minutes
13.
bootup e-lead off
14.
no comfort-noise
15.
no echo-cancel enable
16.
no shutdown
17.
end
DETAILED STEPS
Configuring Polarity and Additional Restrictions on the LMR Voice Port
Perform the following tasks to configure polarity and additional restrictions on the LMR voice port:
•
Configuring Polarity and Additional Restrictions on the M-Lead (optional)
•
Configuring Polarity and Additional Restrictions on the E-Lead (optional)
Configuring Polarity and Additional Restrictions on the M-Lead
Note
Some of the tasks involved in configuring polarity and additional restrictions on the M-lead require different steps for PLAR connections versus connection trunk connections. Be sure to choose the right set of steps for your connection type.
Choose one of the following optional tasks to configure polarity and additional restrictions on the M-lead of an LMR voice port:
•
Configuring the LMR Voice Port to Ignore the M-Lead Signal, Normal Polarity (optional)
•
Configuring the LMR Voice Port to Ignore the M-Lead Signal, Reverse Polarity (optional)
•
Configuring the LMR Voice Port to Gate the M-Lead Audio, Normal Polarity (optional)
•
Configuring the LMR Voice Port to Gate the M-Lead Audio, Reverse Polarity (optional)
•
Configuring the LMR Voice Port to Trigger a Call on First Activity, Normal Polarity (optional)
•
Configuring the LMR Voice Port to Trigger a Call on First Activity, Reverse Polarity (optional)
Configuring the LMR Voice Port to Ignore the M-Lead Signal, Normal Polarity
Perform this task to configure normal polarity on the M-lead and to configure the router to ignore signals sent by voice on the M-lead. The flow of voice packets is determined by VAD. The router sends voice received from the LMR device.
Note
This task applies to PLAR and connection trunk connections. Step 6 uses a different keyword for PLAR and connection trunk. Be sure to choose the correct keyword for your connection type.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr m-lead inactive
5.
define rx-bits idle 0000
6.
define rx-bits seize 1111
or
define rx-bits seize 00007.
exit
8.
dial-peer voice tag voip
9.
vad aggressive
10.
codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr} [bytes payload-size]
11.
end
DETAILED STEPS
Configuring the LMR Voice Port to Ignore the M-Lead Signal, Reverse Polarity
Perform this task to configure reverse polarity on the M-lead and to configure the router to ignore signals sent by voice on the M-lead. The flow of voice packets is determined by VAD. The router sends voice received from the LMR device.
Note
This task applies to PLAR and connection trunk connections. Step 6 uses a different keyword for PLAR and connection trunk. Be sure to choose the correct keyword for your connection type.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr m-lead inactive
5.
define rx-bits idle 1111
6.
define rx-bits seize 0000
or
define rx-bits seize 11117.
exit
8.
dial-peer voice tag voip
9.
vad aggressive
10.
codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr} [bytes payload-size]
11.
end
DETAILED STEPS
Configuring the LMR Voice Port to Gate the M-Lead Audio, Normal Polarity
Perform this task to configure normal polarity on the M-lead and to configure the router to generate VoIP packets when a seize signal is detected on the M-lead. The router stops generating VoIP packets when the seize signal is removed from the M-lead.
Note
This task applies to PLAR and connection trunk connections.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr m-lead audio-gate-in
5.
define rx-bits idle 0000
6.
define rx-bits seize 1111
7.
timing hookflash-input milliseconds
8.
exit
9.
dial-peer voice tag voip
10.
codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr} [bytes payload-size]
11.
end
DETAILED STEPS
Configuring the LMR Voice Port to Gate the M-Lead Audio, Reverse Polarity
Perform this task to configure reverse polarity on the M-lead and to configure the router to generate VoIP packets when a seize signal is detected on the M-lead. The router stops generating VoIP packets when the seize signal is removed from the M-lead.
Note
This task applies to PLAR and connection trunk connections.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr m-lead audio-gate-in
5.
define rx-bits idle 1111
6.
define rx-bits seize 0000
7.
timing hookflash-input milliseconds
8.
exit
9.
dial-peer voice tag voip
10.
no vad aggressive
11.
codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr} [bytes payload-size]
12.
end
DETAILED STEPS
Configuring the LMR Voice Port to Trigger a Call on First Activity, Normal Polarity
Perform this task to configure normal polarity on the M-lead and to configure the router to set up a VoIP connection when the LMR device is not involved in a VoIP connection and the first seize signal is detected on the M-lead. Once the connection is made, the router behaves as in the audio-gate-in option.
Note
This task applies to PLAR connections only.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr m-lead dialin
5.
define rx-bits idle 0000
6.
define rx-bits seize 1111
7.
end
DETAILED STEPS
Configuring the LMR Voice Port to Trigger a Call on First Activity, Reverse Polarity
Perform this task to configure reverse polarity on the M-lead and to configure the router to set up a VoIP connection when the LMR device is not involved in a VoIP connection and the first seize signal is detected on the M-lead. Once the connection is made, the router behaves as in the audio-gate-in option.
Note
This task applies to PLAR connections only.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr m-lead dialin
5.
define rx-bits idle 1111
6.
define rx-bits seize 0000
7.
end
DETAILED STEPS
Configuring Polarity and Additional Restrictions on the E-Lead
Note
Some of the tasks involved in configuring polarity and additional restrictions on the M-lead require different steps for PLAR connections versus connection trunk connections. Be sure to choose the right task for your type of connection.
Choose one of the following optional tasks to configure polarity and additional restrictions on the E-lead of an LMR voice port for PLAR connections:
•
Configuring the E-Lead to Be Always Inactive, Normal Polarity
•
Configuring the E-Lead to Be Always Inactive, Reverse Polarity
•
Configuring the E-Lead for Active Call, Normal Polarity
•
Configuring the E-Lead for Active Call, Reverse Polarity
•
Configuring the E-Lead for Voice Packet, Normal Polarity
•
Configuring the E-Lead for Voice Packet, Reverse Polarity
Configuring the E-Lead to Be Always Inactive, Normal Polarity
Perform this task to configure normal polarity on the E-lead and to configure the router to never send a seize signal on the E-lead to the LMR device. The router sends voice packets to LMR devices.
Note
This task has different steps for PLAR connections and connection trunk connections. Be sure to choose the correct set of steps for your connection type.
Configuring the E-Lead to Be Always Inactive, Normal Polarity, PLAR
Perform this task to configure normal polarity on the E-lead and to configure the router to never send a seize signal on the E-lead to the LMR device for PLAR connections. The router sends voice packets to LMR devices.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr e-lead inactive
5.
define rx-bits idle 0000
6.
define rx-bits seize 1111
7.
end
DETAILED STEPS
Configuring the E-Lead to Be Always Inactive, Normal Polarity, Connection Trunk
Perform this task to configure normal polarity on the E-lead and to configure the router to never send a seize signal on the E-lead to the LMR device for connection trunk connections. The router sends voice packets to LMR devices.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr e-lead inactive
5.
define tx-bits idle 0000
6.
define tx-bits seize 0000
7.
exit
8.
dial-peer voice tag voip
9.
vad aggressive
10.
codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr} [bytes payload-size]
11.
end
DETAILED STEPS
Configuring the E-Lead to Be Always Inactive, Reverse Polarity
Perform this task to configure reverse polarity on the E-lead and to configure the router to never send a seize signal on the E-lead to the LMR device. The router sends voice packets to LMR devices.
Note
This task has different steps for PLAR connections and connection trunk connections. Be sure to choose the correct set of steps for your connection type.
Configuring the E-Lead to Be Always Inactive, Reverse Polarity, PLAR
Perform this task to configure reverse polarity on the E-lead and to configure the router to never send a seize signal on the E-lead to the LMR device for PLAR connections. The router sends voice packets to LMR devices.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr e-lead inactive
5.
define tx-bits idle 1111
6.
define tx-bits seize 0000
7.
end
DETAILED STEPS
Configuring the E-Lead to Be Always Inactive, Reverse Polarity, Connection Trunk
Perform this task to configure reverse polarity on the E-lead and to configure the router to never send a seize signal on the E-lead to the LMR device for connection trunk connections. The router sends voice packets to LMR devices.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr e-lead inactive
5.
define tx-bits idle 1111
6.
define tx-bits seize 1111
7.
exit
8.
dial-peer voice tag voip
9.
vad aggressive
10.
codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr} [bytes payload-size]
11.
end
DETAILED STEPS
Configuring the E-Lead for Active Call, Normal Polarity
Perform this task to configure normal polarity on the E-lead and to configure the router to send a seize signal on the E-lead when the LMR port is connected and removes the seize signal from the E-lead when the LMR port is not involved in a VoIP connection.
Note
This task is optional for PLAR and connection trunk connections. E-lead active call and normal polarity is the default.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr e-lead seize
5.
define rx-bits idle 0000
6.
define rx-bits seize 1111
7.
end
DETAILED STEPS
Configuring the E-Lead for Active Call, Reverse Polarity
Perform this task to configure reverse polarity on the E-lead and to configure the router to send a seize signal on the E-lead when the LMR port is connected and removes the seize signal from the E-lead when the LMR port is not involved in a VoIP connection.
Note
This task applies to PLAR and connection trunk connections.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr e-lead seize
5.
define tx-bits idle 1111
6.
define tx-bits seize 0000
7.
end
DETAILED STEPS
Configuring the E-Lead for Voice Packet, Normal Polarity
Perform this task to configure normal polarity on the E-lead and to configure the router to send a seize signal on the E-lead only when it receives voice packets from the network. When no packets are detected on the network, the seize signal is removed from the E-lead.
Note
This task applies to PLAR and connection trunk connections.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr e-lead voice
5.
define tx-bits idle 0000
6.
define tx-bits seize 1111
7.
timing hangover milliseconds
8.
end
DETAILED STEPS
Configuring the E-Lead for Voice Packet, Reverse Polarity
Perform this task to configure reverse polarity on the E-lead and to configure the router to send a seize signal on the E-lead only when it receives voice packets from the network. When no packets are detected on the network, the seize signal is removed from the E-lead.
Note
This task applies to PLAR and connection trunk connections.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
lmr e-lead voice
5.
define tx-bits idle 1111
6.
define tx-bits seize 0000
7.
timing hangover milliseconds
8.
end
DETAILED STEPS
Troubleshooting Tips
If the E-lead is configured for voice packet and the far-end dial peer uses VAD, the E-lead will turn on and off too frequently causing clipping. Disable VAD on the far-end dial peer to reduce clipping.
Configuring Tone Signaling
To configure a wakeup tone, frequency selection tone, or guard tone to be played out before or with a voice packet, you need to:
•
Create a tone-signal voice class.
•
Configure the desired tones and pauses.
•
Assign the voice class to the LMR voice port.
Note
To avoid voice loss at the receiving end of an LMR system, use the command to configure a delay for the voice packet equal to the sum of the durations of all the injected tones and pauses configured with the command and the inject pause command in this task.
Perform this task to configure tone signaling.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice class tone-signal tag
4.
inject tone index frequency amplitude duration
5.
inject pause index milliseconds
6.
inject guard-tone frequency amplitude [idle]
7.
exit
8.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
9.
voice-class tone-signal tag
10.
exit
11.
end
DETAILED STEPS
Configuring Connections Between LMR Routers
Choose one of the following optional tasks to configure connections between the LMR routers on your IP network. Connection trunk and PLAR are usually used when there are two LMR routers on the network. VoIPmc is used when there are more than two LMR routers on the network.
•
Configuring Connection Trunk (optional)
•
Configuring PLAR (optional)
•
Configuring VoIPmc (optional)
Configuring Connection Trunk
Perform this task to configure connection trunk connections.
Dial Peers
For more information on configuring and troubleshooting dial peers refer to Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
shutdown
5.
connection trunk digits [answer-mode]
6.
no shutdown
7.
exit
8.
dial-peer voice tag pots
9.
destination-pattern [+]string [T]
10.
port {slot-number/subunit-number/port | slot/port:ds0-group-number}
11.
exit
12.
dial-peer voice tag voip
13.
destination-pattern [+]string [T]
14.
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.]host-name | loopback:rtp | loopback:compressed | loopback:uncompressed | ras}
15.
codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr} [bytes payload-size]
16.
dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal] [rtp-nte] [sip-notify]
17.
end
DETAILED STEPS
Configuring PLAR
Perform this task to configure PLAR connections. PLAR connections are activated based on the M-lead, so PLAR can be used only with LMR systems that can raise the M-lead. PLAR connections can also be initiated by the FXS port to dial in to the radio connected to an E&M voice port for point-to-point deployments.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
4.
connection plar digits
5.
no shutdown
6.
end
DETAILED STEPS
Configuring VoIPmc
Perform the tasks in the following sections to configure VoIPmc connections. VoIPmc can be used with connection trunk or PLAR connections. Connection trunk is recommended for E&M voice ports. Connection trunk has a retry mechanism, whereas PLAR does not attempt to retry in case of failure. All VoIPmc configurations require multicast routing and a virtual interface (vif) configured on the router.
•
Configuring Multicast Routing (VoIPmc) (required)
•
Configuring the Virtual Interface (VoIPmc) (required)
•
Configuring VoIP Dial Peers, VoIPmc (required)
•
Configuring E&M Voice Ports, VoIPmc (required)
•
Configuring the Relevant Interface (VoIPmc) (required)
•
Configuring Voice Ports in High-Density Voice Network Modules, VoIPmc (required, if using T1/E1)
Configuring Multicast Routing (VoIPmc)
Perform this task to enable multicast routing.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
ip multicast-routing
4.
end
DETAILED STEPS
Configuring the Virtual Interface (VoIPmc)
Perform this task to configure the virtual interface for multicast fast switching.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
interface type number [name-tag]
4.
ip address ip-address mask
5.
ip pim sparse-dense-mode
6.
end
DETAILED STEPS
Configuring VoIP Dial Peers, VoIPmc
Perform this task to configure the VoIP dial peers on the router.
Dial Peers
For more information on configuring and troubleshooting dial peers refer to Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
dial-peer voice tag voip
4.
destination-pattern [+]string[T]
5.
session protocol multicast
6.
session target ipv4:destination-address
7.
ip precedence number
8.
codec {clear channel | g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r53 | g726r63 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmefr | gsmfr} [bytes payload-size]
9.
end
DETAILED STEPS
Configuring RTP Payload Type
Perform this task to configure RTP payload type on the router.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
dial-peer voice tag voip
4.
rtp payload-type lmr-tone number
5.
rtp payload-type nte-tone number
6.
end
DETAILED STEPS
Configuring E&M Voice Ports, VoIPmc
Perform this task to configure E&M voice ports.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice class permanent tag
4.
signal timing oos timeout [seconds | disabled]
5.
signal keepalive {seconds | disabled}
6.
exit
7.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
8.
voice-class permanent tag
9.
connection trunk digits
10.
music-threshold decibels
11.
operation 4-wire
12.
type {1 | 2 | 3 | 5}
13.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
14.
voice-class permanent tag
15.
connection trunk digits
16.
music-threshold decibels
17.
operation 4-wire
18.
end
DETAILED STEPS
Configuring the Relevant Interface (VoIPmc)
Perform this task to configure either the serial or the Ethernet interface.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
interface type slot/port
4.
ip address ip-address mask [secondary]
5.
ip pim {sparse-mode | dense-mode | sparse-dense-mode}
6.
no shutdown
7.
end
DETAILED STEPS
.
Configuring Voice Ports in High-Density Voice Network Modules, VoIPmc
A multiflex trunk interface card (NM-HDV) in a high-density voice network module requires special voice-port configuration when being connecting for T1/E1 operation. Perform this task to configure a multiflex trunk interface card in a high-density voice network module.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice class permanent tag
4.
signal timing oos timeout [seconds | disabled]
5.
signal keepalive {seconds | disabled}
6.
exit
7.
controller {t1 | e1} slot/port
8.
ds0-group ds0-group-number timeslots timeslot-list type e&m-lmr
9.
exit
10.
voice-port slot/port:ds0-group-number
11.
connection trunk digits [answer-mode]
12.
voice-class permanent tag
13.
end
DETAILED STEPS
Adjusting the Voice Quality on the LMR Voice Port
Perform this task to adjust the voice quality on the LMR voice port.
SUMMARY STEPS
1.
enable
2.
configure terminal
3.
voice class tone-signal tag
4.
digital-filter {1950hz | 2175hz}
5.
exit
6.
voice-port {slot-number/subunit-number/port | slot/port:ds0-group-number}
7.
shutdown
8.
input gain {decibels | auto-control [auto-dbm]}
9.
output attenuation {decibels | auto-control [auto-dbm]}
10.
music-threshold decibels
11.
threshold noise value
12.
voice-class tone-signal tag
13.
no shutdown
14.
exit
15.
voice vad-time milliseconds
16.
end
DETAILED STEPS
Troubleshooting Tips
•
Make sure that the hardware interface is configured correctly for 2-wire or 4-wire operation.
•
Make sure that the hardware interface is configured correctly for E&M signaling Type II, III, or V.
•
Make sure the hardware interface matches the configured transmit and receive signal protocol and signal polarity.
•
Determine whether the problem is related to generic call handling or is LMR-specific.
•
Collect information with debug and show commands.
•
Adjust configuration parameters.
Verifying Land Mobile Radio over IP Enhancement
Perform this task to verify that the Land Mobile Radio over IP Enhancement feature is working.
SUMMARY STEPS
1.
enable
2.
debug vpm signal
3.
debug vpm trunk-sc
4.
show call active voice [[brief] [called-number number | calling-number number] | compact [duration {less time | more time}] | echo-canceller call-id | id identifier | media-inactive [called-number number | calling-number number] | redirect {rtpvt | tbct}]
5.
show interfaces vif 1
6.
show ip mroute [vrf vrf-name] [group-address | group-name] [source-address | source-name] [interface-type interface-number] [summary] [count] [active kbps]
7.
show ip pim [vrf vrf-name] neighbor [interface-type interface-number]
8.
show voice call [[slot/port:ds0-group-number | slot/subunit/port] | status call-id [sample sample-period] | summary]]
9.
show voice dsp
10.
show voice lmr [slot/subunit/port | slot/port:ds0-group-number] [details]
11.
show voice port [[slot/subunit/port | slot/port:ds0-group-number] | summary]
12.
show voip rtp connections [detail]
13.
show voice trunk-conditioning signaling [summary | voice-port]
14.
show voice trunk-conditioning supervisory [summary | voice-port]
15.
test voice port {slot/subunit/port | slot/port:ds0-group-number} detector m-lead {on | off | disable}
16.
test voice port {slot/port:ds0-group-number | slot/subunit/port} relay e-lead {on | off | disable}
17.
test voice port {slot/subunit/port | slot/port:ds0-group-number} inject-tone {local | network} {200hz | 300hz | 500hz | 1000hz | 2000hz | 3000hz | 3200hz | 3400hz | quiet | disable}
DETAILED STEPS
Examples
Examples for commands not shown in this section can be found in the "Configuration Examples for Land Mobile Radio over IP Enhancement" section. The test voice port detector and test voice port relay commands do not produce any output and are not included in this section. This section provides the following output examples:
•
Sample Output for the debug vpm signal Command
•
Sample Output for the debug vpm trunk-sc Command
•
Sample Output for the show call active voice Command
•
Sample Output for the show voice call Command
•
Sample Output for the show voice dsp Command
•
Sample Output for the show voice lmr Command
•
Sample Output for the test voice port inject-tone Command
Sample Output for the debug vpm signal Command
In the following example, the E-lead and M-lead of the LMR voice port are configured as follows:
voice-port 4/0:1lmr m-lead dialinlmr e-lead voiceIn the following sample output of the debug vpm signal command at the terminating side of the call, the output in bold indicates that the call connects:
TermRouter#1w3d:htsp_timer_stop3 htsp_setup_req1w3d:htsp_process_event:[4/0:1(1), LMR_ONHOOK,E_HTSP_SETUP_REQ]lmr_onhook_setup1w3d:htsp_timer_stop htsp_progress1w3d:lmr_start_timer:2000 ms1w3d:htsp_timer - 2000 msechtsp_call_bridged1w3d:htsp_process_event:[4/0:1(1), LMR_WAIT_CUT_THRU,E_HTSP_VOICE_CUT_THROUGH]lmr_cut_thru1w3d:htsp_timer_stop1w3d:lmr_pak_suppress_enable FALSE1w3d:lmr_start_timer2:1800 second1w3d:htsp_timer2 - 1800000 msec1w3d:htsp_process_event:[4/0:1(1), LMR_CONNECT,E_DSP_SIG_0000]lmr_conn_onhook1w3d:htsp_timer_stop1w3d:lmr_start_timer:480 ms1w3d:htsp_timer - 480 msecIn the following sample output of the debug vpm signal command at the originating side of the call, the output in bold indicates that the call connects:
OrigRouter#1w3d:htsp_process_event:[4/0:1(1), LMR_ONHOOK,E_DSP_SIG_1100]lmr_onhook_offok1w3d:htsp_timer_stop htsp_setup_ind1w3d:[4/0:1(1)] get_local_station_id calling num= calling name= callingtime=/18 00:53 orig called=1w3d:htsp_timer - 3000 msec1w3d:htsp_process_event:[4/0:1(1), LMR_WAIT_SETUP_ACK,E_HTSP_SETUP_ACK]lmr_it_setup_ack_get_ack1w3d:htsp_timer_stop1w3d:htsp_process_event:[4/0:1(1), LMR_OFFHOOK, E_HTSP_PROCEEDING]1w3d:htsp_timer_stop3 htsp_setup_reqE_HTSP_VOICE_CUT_THROUGHxsls_waitoff_voice1w3d:htsp_process_event:[4/0:1(1), LMR_OFFHOOK,E_HTSP_VOICE_CUT_THROUGH]lmrffhook_voice_cut1w3d:htsp_timer_stopST3745#1w3d:htsp_process_event:[4/0:1(1), LMR_OFFHOOK,E_HTSP_CONNECT]lmr_offhook_cnect1w3d:htsp_timer_stop1w3d:htsp_timer_stop2Sample Output for the debug vpm trunk-sc Command
When tone is injected into port 1/0/0 locally, the loopback cable sends tone back into the multicast causing the other port 1/0/1 to receive the voice packets and the playout pattern 0xF to appear in the debug vpm trunk-sc command output. In the following output, the signal pattern 0xF shown in bold confirms that voice packet detection is working on the voice port.
*Jun 13 23:52:39.699: 1/0/1: TRUNK_SC state : TRUNK_SC_CONNECT, event TRUNK_VOICE_RCVD*Jun 13 23:52:39.699: 1/0/1: trunk_rtc_gen_pattern : sig pattern 0xFRouter# sh voice call 1/0/01/0/0vtsp level 0 state = S_CONNECTvpm level 1 state = S_TRUNKEDvpm level 0 state = S_UPcalling number , calling name unavailable, calling time 06/13 23:12Router# ***DSP VOICE TX STATISTICS***Tx Vox/Fax Pkts: 12360, Tx Sig Pkts: 0, Tx Comfort Pkts: 7Tx Dur(ms): 2423550, Tx Vox Dur(ms): 247180, Tx Fax Dur(ms): 0Sample Output for the show call active voice Command
The following example shows information from a router in a multicast group for a connection trunk call in progress made in a VoIPmc network:
Telephony call-legs:1SIP call-legs:0H323 call-legs:0MGCP call-legs:0Multicast call-legs:1Total call-legs:2GENERIC:SetupTime=565861590 msIndex=1PeerAddress=PeerSubAddress=PeerId=0PeerIfIndex=0LogicalIfIndex=23ConnectTime=56586159CallDuration=00:00:30 secCallState=4CallOrigin=2ChargedUnits=0InfoType=speechTransmitPackets=148TransmitBytes=24864ReceivePackets=50ReceiveBytes=800TELE:! CoderTypeRate is the codec used in this call! TranslatedCalledNumber is the number being calledConnectionId=[0x467E1D6E 0x5BAD11D8 0x805CDA45 0x64E0FF68] IncomingConnectionId=[0x467E1D6E 0x5BAD11D8 0x805CDA45 0x64E0FF68] CallID=80 TxDuration=30720 ms VoiceTxDuration=2000 ms FaxTxDuration=0 ms CoderTypeRate=g711ulaw NoiseLevel=-72 ACOMLevel=57 OutSignalLevel=-15 InSignalLevel=-71 InfoActivity=2 ERLLevel=57 SessionTarget= ImgPages=0 CallerName= CallerIDBlocked=False OriginalCallingNumber= OriginalCallingOctet=0x0 OriginalCalledNumber= OriginalCalledOctet=0x80 OriginalRedirectCalledNumber= OriginalRedirectCalledOctet=0x0 TranslatedCallingNumber= TranslatedCallingOctet=0x0 TranslatedCalledNumber=7667 TranslatedCalledOctet=0x80 TranslatedRedirectCalledNumber= TranslatedRedirectCalledOctet=0x0GENERIC:SetupTime=565861590 msIndex=2! PeerAddress is the number being calledPeerAddress=7667PeerSubAddress=PeerId=20PeerIfIndex=36LogicalIfIndex=0ConnectTime=0CallDuration=00:00:00 secCallState=4CallOrigin=1ChargedUnits=0InfoType=speechTransmitPackets=0TransmitBytes=0ReceivePackets=148ReceiveBytes=4294965520VOIP:! IP address of the called side and call parametersConnectionId[0x0 0x0 0x0 0x0]IncomingConnectionId[0x467E1D6E 0x5BAD11D8 0x805CDA45 0x64E0FF68] CallID=81 RemoteIPAddress=0.0.0.0 RemoteUDPPort=19878 RemoteSignallingIPAddress=0.0.0.0 RemoteSignallingPort=0 RemoteMediaIPAddress=234.5.6.7 RemoteMediaPort=19878 RoundTripDelay=0 ms SelectedQoS=best-effort tx_DtmfRelay=inband-voice FastConnect=FALSEAnnexE=FALSESeparate H245 Connection=FALSEH245 Tunneling=FALSE! Session protocol and session target set in the selected dial peerSessionProtocol=multicastProtocolCallId=SessionTarget=ipv4:234.5.6.7:19878OnTimeRvPlayout=0GapFillWithSilence=0 msGapFillWithPrediction=0 msGapFillWithInterpolation=0 msGapFillWithRedundancy=0 msHiWaterPlayoutDelay=0 msLoWaterPlayoutDelay=0 msTxPakNumber=100TxSignalPak=0TxComfortNoisePak=1TxDuration=30720TxVoiceDuration=2000RxPakNumber=45RxSignalPak=0RxDuration=0TxVoiceDuration=900VoiceRxDuration=850RxOutOfSeq=0RxLatePak=0RxEarlyPak=0PlayDelayCurrent=64PlayDelayMin=64PlayDelayMax=65PlayDelayClockOffset=11533PlayDelayJitter=67085464PlayErrPredictive=0PlayErrInterpolative=0PlayErrSilence=0PlayErrBufferOverFlow=0PlayErrRetroactive=0PlayErrTalkspurt=0OutSignalLevel=-15InSignalLevel=-71LevelTxPowerMean=0LevelRxPowerMean=-715LevelBgNoise=0ERLLevel=57ACOMLevel=57ErrRxDrop=0ErrTxDrop=0ErrTxControl=45ErrRxControl=66PlayoutMode = undefinedPlayoutInitialDelay=0 msReceiveDelay=0 msLostPackets=0EarlyPackets=0LatePackets=0!VAD status of the callVAD = enabledCoderTypeRate=g711ulawCodecBytes=0Media Setting=flow-aroundCallerName=CallerIDBlocked=FalseOriginalCallingNumber=OriginalCallingOctet=0x0OriginalCalledNumber=OriginalCalledOctet=0x0OriginalRedirectCalledNumber=OriginalRedirectCalledOctet=0x0TranslatedCallingNumber=TranslatedCallingOctet=0x0TranslatedCalledNumber=TranslatedCalledOctet=0x0TranslatedRedirectCalledNumber= TranslatedRedirectCalledOctet=0x0 MediaInactiveDetected=no MediaInactiveTimestamp= MediaControlReceived= Username= Telephony call-legs:1 SIP call-legs:0 H323 call-legs:0 MGCP call-legs:0 Multicast call-legs:1 Total call-legs:2Sample Output for the show voice call Command
In the following example, the fields in bold show that the call is connected:
1/0/0vtsp level 0 state = S_CONNECTvpm level 1 state = S_TRUNKEDvpm level 0 state = S_UPcalling number , calling name unavailable, calling time 06/13 23:12Router# ***DSP VOICE TX STATISTICS***Tx Vox/Fax Pkts: 5022, Tx Sig Pkts: 0, Tx Comfort Pkts: 4Tx Dur(ms): 2182560, Tx Vox Dur(ms): 100430, Tx Fax Dur(ms): 0Sample Output for the show voice dsp Command
When nothing is connected to the E&M voice port, there is a very slow increment of the PACK COUNT field due to RTP Control Protocol (RTCP). However, when the radio is transmitting or receiving, the PACK COUNT increments by a relatively large amount. Incrementing of the PACK COUNT field also indicates whether the packets are being received and sent by the router. The output is shown for the C542 and C5510 DSP.
Router# show voice dsp!The following output is for the NM-HD-2V analog moduleDSP DSP DSPWARE CURR BOOT PAK TX/RXTYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT==== === == ======== ======= ===== ======= === == ========= == ===== ===========C542 001 01 g711ulaw 4.3.10 busy idle 0 0 1/0/0 NA 0 16/16C542 002 01 g711ulaw 4.3.10 busy idle 0 0 1/0/1 NA 0 16/16!The following output is for the NM-HD-2V analog module*DSP SIGNALING CHANNELS*DSP DSP DSPWARE CURR BOOT PAK TX/RXTYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT===== === == ======== ======= ===== ======= === == ========= == ==== ===========C5510 001 01 {flex} 4.3.10 alloc idle 0 0 2/1/0 02 0 0/0C5510 001 02 {flex} 4.3.10 alloc idle 0 0 2/1/1 02 0 0/0Sample Output for the show voice lmr Command
Use the show voice lmr command to verify LMR-specific configuration, for example, E-lead and M-lead status, timeouts, and injected tones, shown in bold in the following example output.
Router# show voice lmr 2/0/0 details2/0/0=========Description:Connection type: n/aOut Attenuation = 0 db, In Gain = 0 dBTiming hangover: 500 msE-lead capability is inactive, polarity = normalM-lead capability is inactive, polarity = normalTiming hookflash-in: 480Timing delay-voice: 470 msMusic On Hold Threshold: -38 dB, Noise Threshold: -62 dBE&M type: 1, Operation: 2-wireImpedance is set to 600r Ohmlmr tear down timeout is set to 1800 secondlmr PTT transmit timeout is not setlmr PTT receive timeout is not setvoice-class tone-signal testinject tone 1 1950 3 150inject tone 2 2000 0 60inject pause 3 60inject tone 4 2175 3 150inject tone 5 1000 0 50inject guard-tone 1950 -10state = LMR_CONNECT, e-lead = off, m-lead = offfull duplex, voice path = rxTerminating side of the connectionTransmitPackets=113, TransmitBytes=2241ReceivePackets=113, ReceiveBytes=2241CoderTypeRate=g729r8NoiseLevel=-66, ACOMLevel=22OutSignalLevel=-68, InSignalLevel=-79PeerAddress=37200PeerSubAddress=PeerId=200SessionTarget=RemoteIPAddress=10.5.25.40, RemoteUDPPort=17272Remote SignallingIPAddress=10.5.25.40, Port=15418Remote MediaIPAddress=10.5.25.40, Port=17272RoundTripDelay=0 msSessionProtocol=ciscoVAD =enabledSelectedQoS=best-effortProtocolCallId=SessionTarget=Sample Output for the test voice port inject-tone Command
When packets are sent to the multicast group using the test voice port inject-tone network command, the packets are also received on the ports. The same multicast IP:port is used for sending and receiving when configured for multicast. Only symmetric send and receive configuration is supported on dial peers.
Router# test voice port 1/0/1 inject-tone network 1000*Jun 13 23:44:51.943: 1/0/0: TRUNK_SC state : TRUNK_SC_CONNECT, event TRUNK_VOICE_RCVD*Jun 13 23:44:51.943: 1/0/0: trunk_rtc_gen_pattern : sig pattern 0xF*Jun 13 23:44:52.035: 1/0/1: TRUNK_SC state : TRUNK_SC_CONNECT, event TRUNK_VOICE_RCVD*Jun 13 23:44:52.035: 1/0/1: trunk_rtc_gen_pattern : sig pattern 0xFRouter# test voice port 1/0/1 inject-tone network quiet*Jun 13 23:45:07.455: 1/0/0: TRUNK_SC state : TRUNK_SC_CONNECT, event TRUNK_VOICE_STOPPED*Jun 13 23:45:07.455: 1/0/0: trunk_rtc_gen_pattern : sig pattern 0x0*Jun 13 23:45:07.543: 1/0/1: TRUNK_SC state : TRUNK_SC_CONNECT, event TRUNK_VOICE_STOPPED*Jun 13 23:45:07.543: 1/0/1: trunk_rtc_gen_pattern : sig pattern 0x0Injecting the tone locally into the E&M port causes the high dB levels in TX and RX, which confirms that the audio leads of the E&M voice port are functional. The tone remains active for only 30 seconds.
Router# test voice port 1/0/0 inject-tone local 1000*Jun 13 23:52:39.699: 1/0/1: TRUNK_SC state : TRUNK_SC_CONNECT, event TRUNK_VOICE_RCVD*Jun 13 23:52:39.699: 1/0/1: trunk_rtc_gen_pattern : sig pattern 0xFRouter# sh voice call 1/0/01/0/0vtsp level 0 state = S_CONNECTvpm level 1 state = S_TRUNKEDvpm level 0 state = S_UPcalling number , calling name unavailable, calling time 06/13 23:12Router# ***DSP VOICE TX STATISTICS***Tx Vox/Fax Pkts: 12360, Tx Sig Pkts: 0, Tx Comfort Pkts: 7Tx Dur(ms): 2423550, Tx Vox Dur(ms): 247180, Tx Fax Dur(ms): 0<<<<Removed for clarity >>>>>***DSP LEVELS***TDM Bus Levels(dBm0): Rx -0.2 from PBX/Phone, Tx -0.5 to PBX/PhoneTDM ACOM Levels(dBm0): +20.0, TDM ERL Level(dBm0): +20.0TDM Bgd Levels(dBm0): -75.2, with activity being voice***DSP VOICE ERROR STATISTICS***Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0Configuration Examples for Land Mobile Radio over IP Enhancement
This section provides configuration examples to match the identified configuration tasks in the previous section. This section does not provide examples for every option under every configuration task.
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Configuring Connection Trunk on an Analog LMR Voice Port: Example
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Verifying Connection Trunk on an Analog LMR Voice Port: Example
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Configuring Connection Trunk on a Digital LMR Voice Port: Example
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Verifying Connection Trunk on a Digital LMR Voice Port: Example
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Configuring PLAR on an Analog LMR Voice Port: Example
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Verifying PLAR on an Analog LMR Voice Port: Example
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Configuring PLAR on a Digital LMR Voice Port: Example
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Verifying PLAR on a Digital LMR Voice Port: Example
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Configuring VoIPmc with Connection Trunk on an Analog LMR Voice Port: Example
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Verifying VoIPmc with Connection Trunk on an Analog LMR Voice Port: Example
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Configuring VoIPmc with Connection Trunk on a Digital LMR Voice Port: Example
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Verifying VoIPmc with Connection Trunk on a Digital LMR Voice Port: Example
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Configuring VoIPmc with Connection PLAR on an Analog LMR Voice Port: Example
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Verifying VoIPmc with Connection PLAR on an Analog LMR Voice Port: Example
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Configuring VoIPmc with Connection PLAR on a Digital LMR Voice Port: Example
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Verifying VoIPmc with Connection PLAR on a Digital LMR Voice Port: Example
Configuring Connection Trunk on an Analog LMR Voice Port: Example
The following output shows an analog LMR voice port and a connection trunk connection configured on the router:
Router# show running-configBuilding configuration...Current configuration : 2456 bytes!version 12.3service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router!!!interface FastEthernet0/0ip address 10.2.82.81 255.255.255.0load-interval 30duplex autospeed auto!!ip classlessip route 10.2.82.0 255.255.255.0 FastEthernet0/0!!voice-port 1/0/0lmr m-lead audio-gate-inlmr e-lead voiceoperation 4-wiretype 5signal lmrtimeouts call-disconnect 3connection trunk 7667!!dial-peer voice 30 voipdestination-pattern 7667session target ipv4:10.2.82.82codec g711ulawvad aggressive!dial-peer voice 40 potsdestination-pattern 8668port 1/0/0!Verifying Connection Trunk on an Analog LMR Voice Port: Example
In this example, the show voice port summary command displays configuration information about the LMR voice port. The fields in bold show that the voice port's signaling type is E&M LMR and that connection trunk is configured.
Router# show voice port summaryIN OUTPORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC========= == ============ ===== ==== ======== ======== ==1/0/0 -- e&m-lmr up up trunked trunked yIn this example, the show voice trunk-conditioning supervisory command displays the status of trunk supervision and configuration parameters for the LMR voice port. The voice packet detection enable statement shows that the voice option of the command has been configured. If an option other than voice has been configured, the voice packet detection disable statement appears.
Router# show voice trunk-conditioning supervisorySLOW SCAN, SCAN LMR1/0/0 : state : TRUNK_SC_CONN_WO_CLASS, voice : off , signal : on ,masterstatus: rcv IDLE, trunk connectedsequence oos : no-actionpattern :tx_idle = 0000timing : idle = 480, restart = 0, standby = 0, timeout = 30supp_all = 0, supp_voice = 0, keep_alive = 5timer: oos_ais_timer = 0, timer = 0voice packet detection enableIn this example, the show voice trunk-conditioning signaling command displays the status of trunk-conditioning signaling and timing parameters for the LMR voice port.
When the trunk is the ACTIVE state, the forced playout pattern field is set to 0xF. In this example, the forced playout pattern field is set to 0x0 indicating that the trunk is idle.
Router# show voice trunk-conditioning signaling1/0/0 :hardware-state IDLE signal type is NorthamericanCASstatus : IDLEforced playout pattern = 0x0last-TX-ABCD=0000, last-RX-ABCD=0000idle monitoring : txtx_idle = TRUE, rx_idle = FALSE, tx_oos = FALSE, lost_keepalive = FALSEtrunk_down_timer = 0, rx_ais_duration = 0, idle_timer = 0,tx_oos_timer = 0In this example, the show voice call summary command displays the call status for the LMR voice port:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE============== ======== === ==================== ======================1/0/0 g711ulaw y S_CONNECT S_TRUNKEDConfiguring Connection Trunk on a Digital LMR Voice Port: Example
The following output shows a digital LMR voice port and a connection trunk connection configured on the router:
Router# show running-configBuilding configuration...Current configuration : 1615 bytes!version 12.3service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router!!voice-card 1dspfarm!!controller T1 1/0framing esfcrc-threshold 320clock source internallinecode b8zsds0-group 1 timeslots 1 type e&m-lmr!!interface FastEthernet0/0ip address 10.2.82.82 255.255.255.0load-interval 30duplex autospeed auto!!ip classlessip route 10.2.82.0 255.255.255.0 FastEthernet0/0!!!voice-port 1/0:1lmr m-lead audio-gate-intimeouts call-disconnect 3connection trunk 8668!!dial-peer voice 30 voipdestination-pattern 8668session target ipv4:10.2.82.81codec g711ulawvad aggressive!dial-peer voice 40 potsdestination-pattern 7667port 1/0:1!Verifying Connection Trunk on a Digital LMR Voice Port: Example
In this example, the show voice call summary command displays the call status for the LMR voice port:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE============== ======== === ==================== ======================1/0:1.1 g711ulaw y S_CONNECT S_TRUNKEDIn this example, the show voice trunk-conditioning supervisory command displays the status of trunk supervision and configuration parameters for the LMR voice port:
Router# show voice trunk-conditioning supervisorySLOW SCAN, SCAN LMR1/0:1(1) : state : TRUNK_SC_CONN_WO_CLASS, voice : off , signal : on ,masterstatus: rcv IDLE, trunk connectedsequence oos : no-actionpattern :tx_idle = 0000timing : idle = 480, restart = 0, standby = 0, timeout = 30supp_all = 0, supp_voice = 0, keep_alive = 5timer: oos_ais_timer = 0, timer = 0voice packet detection disableIn this example, the show voice trunk-conditioning signaling command displays the status of trunk-conditioning signaling and timing parameters for the LMR voice port. When the trunk is the ACTIVE state, the forced playout pattern field is set to 0xF.
Router# show voice trunk-conditioning signaling1/0:1(1) :hardware-state IDLE signal type is NorthamericanCASstatus : IDLEforced playout pattern = STOPPEDlast-TX-ABCD=0000, last-RX-ABCD=0000idle monitoring : txtx_idle = TRUE, rx_idle = FALSE, tx_oos = FALSE, lost_keepalive = FALSEtrunk_down_timer = 0, rx_ais_duration = 0, idle_timer = 0,tx_oos_timer = 0In this example, the show voice port summary command displays configuration information about the LMR voice port:
Router# show voice port summaryIN OUTPORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC========= == ============ ===== ==== ======== ======== ==1/0:1 01 e&m-lmr up up trunked trunked yConfiguring PLAR on an Analog LMR Voice Port: Example
The following output shows an analog LMR voice port and a PLAR connection configured on the router:
Router# show running-configBuilding configuration...Current configuration : 2427 bytes!version 12.3service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router!!interface FastEthernet0/0ip address 10.2.82.81 255.255.255.0load-interval 30duplex autospeed auto!!ip classlessip route 10.2.82.0 255.255.255.0 FastEthernet0/0!!voice-port 1/0/0lmr m-lead audio-gate-inlmr e-lead voiceoperation 4-wiretype 5signal lmrconnection plar 7667!!!dial-peer voice 30 voipdestination-pattern 7667session target ipv4:10.2.82.82codec g711ulawvad aggressive!dial-peer voice 40 potsdestination-pattern 8668port 1/0/0!Verifying PLAR on an Analog LMR Voice Port: Example
In this example, the show voice port summary command displays configuration information about the LMR voice port:
Router# show voice port summaryIN OUTPORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC========= == ============ ===== ==== ======== ======== ==1/0/0 -- e&m-lmr up up idle idle yIn this example, the show voice call summary command displays the call status for the LMR voice port:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE============== ======== === ==================== ======================1/0/0 g711ulaw y S_CONNECT LMR_CONNECTIn this example, the show voip rtp connections command displays the local and remote calling ID and IP address and port information for active RTP connections:
Router# show voip rtp connectionsVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 67 68 16884 16572 10.2.82.81 10.2.82.82Found 1 active RTP connectionsConfiguring PLAR on a Digital LMR Voice Port: Example
The following output shows a digital LMR voice port and a PLAR connection configured on the router:
Router# show running-configBuilding configuration...Current configuration : 1579 bytes!version 12.3service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router!!voice-card 1dspfarm!!controller T1 1/0framing esfcrc-threshold 320clock source internallinecode b8zsds0-group 1 timeslots 1 type e&m-lmr!!interface FastEthernet0/0ip address 10.2.82.82 255.255.255.0load-interval 30duplex autospeed auto!!ip classlessip route 10.2.82.0 255.255.255.0 FastEthernet0/0!!voice-port 1/0:1lmr m-lead dialinconnection plar 8668!!dial-peer voice 30 voipdestination-pattern 8668session target ipv4:10.2.82.81codec g711ulawvad aggressive!dial-peer voice 40 potsdestination-pattern 7667port 1/0:1Verifying PLAR on a Digital LMR Voice Port: Example
In this example, the show voice call summary command displays the call status for the LMR voice port:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE============== ======== === ==================== ======================1/0:1.1 g711ulaw y S_CONNECT LMR_CONNECTIn this example, the show voice port summary command displays configuration information about the LMR voice port:
Router# show voice port summaryIN OUTPORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC========= == ============ ===== ==== ======== ======== ==1/0:1 01 e&m-lmr up up idle seized yIn this example, the show voip rtp connections command displays the local and remote calling ID and IP address and port information for active RTP connections:
Router# show voip rtp connectionsVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 80 79 16572 16884 10.2.82.82 10.2.82.81Found 1 active RTP connectionsConfiguring VoIPmc with Connection Trunk on an Analog LMR Voice Port: Example
The following output shows an analog LMR voice port and a VoIPmc connection using connection trunk configured on the router:
Router# show running-configBuilding configuration...Current configuration : 3101 bytes!version 12.3service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router!!voice class permanent 1signal timing oos timeout disabledsignal keepalive disabled!!!interface Loopback1ip address 10.0.0.1 255.0.0.0ip pim sparse-dense-mode!interface Vif1ip address 10.3.92.1 255.255.255.0ip accounting output-packetsip pim sparse-dense-modeload-interval 30!interface FastEthernet0/0ip address 10.2.82.81 255.255.255.0ip pim sparse-dense-modeload-interval 30duplex autospeed auto!!ip classlessip route 10.0.0.0 255.0.0.0 FastEthernet0/0ip route 10.2.82.0 255.255.255.0 FastEthernet0/0ip route 10.2.92.0 255.255.255.0 FastEthernet0/0!ip http serverno ip http secure-serverip pim bidir-enableip pim rp-address 10.0.0.1 override bidirip pim send-rp-announce Loopback1 scope 16ip pim send-rp-discovery Loopback1 scope 16!voice-port 1/0/0voice-class permanent 1lmr m-lead audio-gate-inlmr e-lead voiceoperation 4-wiretype 5signal lmrtimeouts call-disconnect 3connection trunk 7667!!dial-peer voice 20 voipdestination-pattern 7667session protocol multicastsession target ipv4:239.5.6.7:19878codec g711ulawvad aggressive!!Verifying VoIPmc with Connection Trunk on an Analog LMR Voice Port: Example
In this example, the show voice port summary command displays configuration information about the LMR voice port:
Router# show voice port summaryIN OUTPORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC========= == ============ ===== ==== ======== ======== ==1/0/0 -- e&m-lmr up up trunked trunked yIn this example, the show voice trunk-conditioning supervisory command displays the status of trunk supervision and configuration parameters for the LMR voice port. The voice packet detection enable statement shows that the voice option of the command has been configured. If an option other than voice has been configured, the voice packet detection disable statement appears.
Router# show voice trunk-conditioning supervisorySLOW SCAN, SCAN LMR1/0/0 : state : TRUNK_SC_CONNECT, voice : off , signal : on ,masterstatus: rcv IDLE, trunk connectedsequence oos : idle-onlypattern :rx_idle = 0000 tx_idle = 0000timing : idle = 480, restart = 0, standby = 0, timeout = 0supp_all = 0, supp_voice = 0, keep_alive = 0timer: oos_ais_timer = 0, timer = 0voice packet detection enableIn this example, the show voice trunk-conditioning signaling command displays the status of trunk-conditioning signaling and timing parameters for the LMR voice port. When the trunk is the ACTIVE state, the forced playout pattern field is set to 0xF.
Router# show voice trunk-conditioning signaling1/0/0 :hardware-state IDLE signal type is NorthamericanCASstatus : IDLEforced playout pattern = 0x0last-TX-ABCD=0000, last-RX-ABCD=0000idle monitoring : txtx_idle = TRUE, rx_idle = FALSE, tx_oos = FALSE, lost_keepalive = FALSEtrunk_down_timer = 0, rx_ais_duration = 0, idle_timer = 0,tx_oos_timer = 0In this example, the show voice call summary command displays the call status for the LMR voice port:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE============== ======== === ==================== ======================1/0/0 g711ulaw y S_CONNECT S_TRUNKEDIn this example, the show ip pim neighbor command lists the PIM neighbors discovered by the Cisco IOS software:
Router# show ip pim neighborPIM Neighbor TableNeighbor Interface Uptime/Expires Ver DRAddress Prio/Mode10.2.82.82 FastEthernet0/0 00:06:51/00:01:17 v2 1 / DR B SIn this example, the show voip rtp connections command displays the local and remote calling ID and IP address and port information for active RTP connections:
Router# show voip rtp connectionsVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 37 36 17090 19878 10.3.92.2 239.5.6.7Found 1 active RTP connectionsIn this example, the show interfaces vif 1 command displays statistics for the vif 1 interface:
Router# show interfaces vif 1Vif1 is up, line protocol is upHardware is PGMVIFInternet address is 92.3.92.1/24MTU 1514 bytes, BW 8000000 Kbit, DLY 5000 usec,reliability 255/255, txload 1/255, rxload 1/255Encapsulation LOOPBACK, loopback not setLast input 00:00:02, output never, output hang neverLast clearing of "show interface" counters 00:10:57Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0Queueing strategy: fifoOutput queue: 0/0 (size/max)30 second input rate 0 bits/sec, 0 packets/sec30 second output rate 0 bits/sec, 0 packets/sec181 packets input, 29721 bytes, 0 no bufferReceived 0 broadcasts, 0 runts, 0 giants, 0 throttles0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored, 0 abort482 packets output, 80478 bytes, 0 underruns0 output errors, 0 collisions, 0 interface resets0 output buffer failures, 0 output buffers swapped outConfiguring VoIPmc with Connection Trunk on a Digital LMR Voice Port: Example
The following output shows a digital LMR voice port and a VoIPmc connection using connection trunk configured on the router:
Router# show running-configBuilding configuration...Current configuration : 2250 bytes!version 12.3service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router!!voice-card 1dspfarm!!voice class permanent 1signal timing oos timeout disabledsignal keepalive disabled!!controller T1 1/0framing esfcrc-threshold 320clock source internallinecode b8zsds0-group 1 timeslots 1 type e&m-lmr!!interface Loopback1ip address 10.0.0.1 255.0.0.0ip pim sparse-dense-mode!interface Vif1ip address 10.2.92.1 255.255.255.0ip accounting output-packetsip pim sparse-dense-modeload-interval 30!interface FastEthernet0/0ip address 10.2.82.82 255.255.255.0ip pim sparse-dense-modeload-interval 30duplex autospeed auto!!ip classlessip route 10.0.0.0 255.0.0.0 FastEthernet0/0ip route 10.2.82.0 255.255.255.0 FastEthernet0/0ip route 10.3.92.0 255.255.255.0 FastEthernet0/0!ip http serverno ip http secure-serverip pim bidir-enableip pim rp-address 50.0.0.1 override bidirip pim send-rp-announce Loopback1 scope 16ip pim send-rp-discovery scope 16!!voice-port 1/0:1voice-class permanent 1lmr m-lead audio-gate-intimeouts call-disconnect 3connection trunk 8668!dial-peer voice 20 voipdestination-pattern 8668session protocol multicastsession target ipv4:239.5.6.7:19878codec g711ulawvad aggressive!Verifying VoIPmc with Connection Trunk on a Digital LMR Voice Port: Example
In this example, the show voice port summary command displays configuration information about the LMR voice port:
Router# show voice port summaryIN OUTPORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC========= == ============ ===== ==== ======== ======== ==1/0:1 01 e&m-lmr up up trunked trunked yIn this example, the show voice trunk-conditioning supervisory command displays the status of trunk supervision and configuration parameters for the LMR voice port. The voice packet detection disable statement shows that an option other than voice is configured for the command. If the voice option of the lmr e-lead command was configured, the voice packet detection enable statement would appear.
Router# show voice trunk-conditioning supervisorySLOW SCAN, SCAN LMR1/0:1(1) : state : TRUNK_SC_CONNECT, voice : off , signal : on ,masterstatus: rcv IDLE, trunk connectedsequence oos : no-actionpattern :tx_idle = 0000timing : idle = 480, restart = 0, standby = 0, timeout = 0supp_all = 0, supp_voice = 0, keep_alive = 0timer: oos_ais_timer = 0, timer = 0voice packet detction disableIn this example, the show voice trunk-conditioning signaling command displays the status of trunk-conditioning signaling and timing parameters for the LMR voice port. When the trunk is the ACTIVE state, the forced playout pattern field is set to 0xF.
Router# show voice trunk-conditioning signaling1/0:1(1) :hardware-state IDLE signal type is NorthamericanCASstatus : IDLEforced playout pattern = STOPPEDlast-TX-ABCD=0000, last-RX-ABCD=0000idle monitoring : txtx_idle = TRUE, rx_idle = FALSE, tx_oos = FALSE, lost_keepalive = FALSEtrunk_down_timer = 0, rx_ais_duration = 0, idle_timer = 0,tx_oos_timer = 0In this example, the show voice call summary command displays the call status for the LMR voice port:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE============== ======== === ==================== ======================1/0:1.1 g711ulaw y S_CONNECT S_TRUNKEDIn this example, the show ip pim neighbor command lists the PIM neighbors discovered by the Cisco IOS software:
Router# show ip pim neighborPIM Neighbor TableNeighbor Interface Uptime/Expires Ver DRAddress Prio/Mode10.2.82.81 FastEthernet0/0 00:25:20/00:01:34 v2 1 / B SConfiguring VoIPmc with Connection PLAR on an Analog LMR Voice Port: Example
The following output shows an analog LMR voice port and a VoIPmc connection using PLAR configured on the router:
Router# show running-configBuilding configuration...Current configuration : 3065 bytes!version 12.3service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router!!voice class permanent 1signal timing oos timeout disabledsignal keepalive disabled!!interface Loopback1ip address 10.0.0.1 255.0.0.0ip pim sparse-dense-mode!interface Vif1ip address 10.3.92.1 255.255.255.0ip accounting output-packetsip pim sparse-dense-modeload-interval 30!interface FastEthernet0/0ip address 10.2.82.81 255.255.255.0ip pim sparse-dense-modeload-interval 30duplex autospeed auto!!ip classlessip route 10.0.0.0 255.0.0.0 FastEthernet0/0ip route 10.2.82.0 255.255.255.0 FastEthernet0/0ip route 10.2.92.0 255.255.255.0 FastEthernet0/0!ip http serverno ip http secure-serverip pim bidir-enableip pim rp-address 10.0.0.1 override bidirip pim send-rp-announce Loopback1 scope 16ip pim send-rp-discovery Loopback1 scope 16!!voice-port 1/0/0voice-class permanent 1lmr m-lead dialinlmr e-lead voiceoperation 4-wiretype 5signal lmrconnection plar 7667!!dial-peer voice 20 voipdestination-pattern 7667session protocol multicastsession target ipv4:239.5.6.7:19878codec g711ulawvad aggressive!Verifying VoIPmc with Connection PLAR on an Analog LMR Voice Port: Example
In this example, the show voice port summary command displays configuration information about the LMR voice port:
Router# show voice port summaryIN OUTPORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC========= == ============ ===== ==== ======== ======== ==1/0/0 -- e&m-lmr up up seized seized yIn this example, the show voice call summary command displays the call status for the LMR voice port:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE============== ======== === ==================== ======================1/0/0 g711ulaw y S_CONNECT LMR_CONNECTIn this example, the show voip rtp connections command displays the local and remote calling ID and IP address and port information for active RTP connections:
Router# show voip rtp connectionsVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 45 44 25470 19878 10.3.92.2 239.5.6.7Found 1 active RTP connectionsIn this example, the show ip pim neighbor command lists the PIM neighbors discovered by the Cisco IOS software:
Router# show ip pim neighborPIM Neighbor TableNeighbor Interface Uptime/Expires Ver DRAddress Prio/Mode10.2.82.82 FastEthernet0/0 00:07:20/00:01:18 v2 1 / DR B SConfiguring VoIPmc with Connection PLAR on a Digital LMR Voice Port: Example
The following output shows a digital LMR voice port and a VoIPmc connection using PLAR configured on the router:
Router# show running-configBuilding configuration...Current configuration : 2214 bytes!version 12.3service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router!!voice-card 1dspfarm!!voice class permanent 1signal timing oos timeout disabledsignal keepalive disabled!!controller T1 1/0framing esfcrc-threshold 320clock source internallinecode b8zsds0-group 1 timeslots 1 type e&m-lmr!!interface Loopback1ip address 10.0.0.1 255.0.0.0ip pim sparse-dense-mode!interface Vif1ip address 10.2.92.1 255.255.255.0ip accounting output-packetsip pim sparse-dense-modeload-interval 30!interface FastEthernet0/0ip address 10.2.82.82 255.255.255.0ip pim sparse-dense-modeload-interval 30duplex autospeed auto!!ip classlessip route 10.0.0.0 255.0.0.0 FastEthernet0/0ip route 10.2.82.0 255.255.255.0 FastEthernet0/0ip route 10.3.92.0 255.255.255.0 FastEthernet0/0!ip http serverno ip http secure-serverip pim bidir-enableip pim rp-address 10.0.0.1 override bidirip pim send-rp-announce Loopback1 scope 16ip pim send-rp-discovery scope 16!!voice-port 1/0:1voice-class permanent 1lmr m-lead dialinconnection plar 8668!!dial-peer voice 20 voipdestination-pattern 8668session protocol multicastsession target ipv4:239.5.6.7:19878codec g711ulawvad aggressive!Verifying VoIPmc with Connection PLAR on a Digital LMR Voice Port: Example
In this example, the show voip rtp connections command displays the local and remote calling ID and IP address and port information for active RTP connections:
Router# show voip rtp connectionsVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 58 57 29636 19878 10.2.92.2 239.5.6.7Found 1 active RTP connectionsIn this example, the show voice call summary command displays the call status for the LMR voice port:
Router# show voice call summaryPORT CODEC VAD VTSP STATE VPM STATE============== ======== === ==================== ======================1/0:1.1 g711ulaw y S_CONNECT LMR_CONNECTIn this example, the show voice port summary command displays configuration information about the LMR voice port:
Router# show voice port summaryIN OUTPORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC========= == ============ ===== ==== ======== ======== ==1/0:1 01 e&m-lmr up up idle seized yIn this example, the show ip pim neighbor command lists the PIM neighbors discovered by the Cisco IOS software:
Router# show ip pim neighborPIM Neighbor TableNeighbor Interface Uptime/Expires Ver DRAddress Prio/Mode10.2.82.81 FastEthernet0/0 00:07:13/00:01:25 v2 1 / B SIn this example, the show ip mroute summary command displays the contents of the IP multicast (mroute) routing table:
Router# show ip mroute summaryIP Multicast Routing TableFlags: D - Dense, S - Sparse, B - Bidir Group, s - SSM Group, C - Connected,L - Local, P - Pruned, R - RP-bit set, F - Register flag,T - SPT-bit set, J - Join SPT, M - MSDP created entry,X - Proxy Join Timer Running, A - Candidate for MSDP Advertisement,U - URD, I - Received Source Specific Host Report, Z - Multicast TunnelY - Joined MDT-data group, y - Sending to MDT-data groupOutgoing interface flags: H - Hardware switchedTimers: Uptime/ExpiresInterface state: Interface, Next-Hop or VCD, State/Mode(*, 239.5.6.7), 00:01:56/00:02:53, RP 50.0.0.1, OIF count: 2, flags: BC(*, 239.0.1.39), 00:08:03/00:02:56, RP 50.0.0.1, OIF count: 3, flags: BCL(*, 239.0.1.40), 00:08:08/00:02:55, RP 50.0.0.1, OIF count: 2, flags: BCLAdditional References
The following sections provide references related to the Land Mobile Radio over IP Enhancement feature:
Related Documents
Related Topic Document TitleCisco IOS commands
Cisco IOS Voice Command Reference, Release 12.3 T
Codecs
Tech Note, Understanding Codecs: Complexity, Hardware Support, MOS, and Negotiation, Document ID 14069
E&M interfaces
•
Tech Note, Voice - Analog E&M Signaling Overview, Document ID 14003
•
Tech Note, Understanding and Troubleshooting Analog E & M Interface Types and Wiring Arrangements, Document ID 8111
Land Mobile Radio over IP
Cisco Land Mobile Radio over IP Solution Reference Network Design
V3PN
White Paper, Voice and Video Enabled IPSec VPN (V3PN)
VoIPmc
Standards
Standards TitleNo new or modified standards are supported by this feature, and support for existing standards has not been modified by this feature.
—
MIBs
RFCs
Technical Assistance
Command Reference
This section documents only new and modified commands.
New Commands
Modified Commands
bootup e-lead off
To prevent an analog ear and mouth (E&M) voice port from keying the attached radio on router boot up, use the bootup e-lead off command in voice-port configuration mode. To allow the analog E&M voice port to key the attached radio on boot up, use the no form of this command.
bootup e-lead off
no bootup e-lead off
Syntax Description
This command has no arguments or keywords.
Defaults
The analog E&M voice port keys the attached radio on radio boot up.
Command Modes
Voice-port configuration
Command History
Usage Guidelines
This command configures the E-lead behavior on boot up for both voice ports on the voice interface card (VIC).
Examples
The following example configures the analog E&M voice port to not key the attached radio on router boot up:
voice-port 1/0/0bootup e-lead offdefine
To define the transmit and receive bits for North American ear and mouth (E&M), E&M Mercury Exchange Limited Channel-Associated Signaling (MELCAS), and Land Mobile Radio (LMR) voice signaling, use the define command in voice-port configuration mode. To restore the default value, use the no form of this command.
define {tx-bits | rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 | 0110 | 0111 | 1000 | 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}
no define {tx-bits | rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 | 0110 | 0111 | 1000 | 1001 | 1010 | 1011 | 1100 | 1101 | 1110 | 1111}
Syntax Description
Defaults
The default is to use the preset signaling patterns as defined in American National Standards Institute (ANSI) and European Conference of Postal and Telecommunications Administrations (CEPT) standards, as follows:
•
For North American E&M:
–
tx-bits idle 0000 (0001 if on E1 trunk)
–
tx-bits seize 1111
–
rx-bits idle 0000
–
rx-bits seize 1111
•
For E&M MELCAS:
–
tx-bits idle 1101
–
tx-bits seize 0101
–
rx-bits idle 1101
–
rx-bits seize 0101
•
For LMR:
–
tx-bits idle 0000
–
tx-bits seize 1111
–
rx-bits idle 0000
–
rx-bits seize 1111
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The define command applies to E&M digital voice ports associated with T1/E1 controllers.
Use the define command to match the E&M bit patterns with the attached telephony device. Be careful not to define invalid configurations, such as all 0000 on E1, or identical seized and idle states. Use this command with the ignore command.
In LMR signaling, the define command is used to define polarity on E&M analog and digital voice ports.
Examples
To configure a voice port on a Cisco 2600 or Cisco 3600 series router that is sending traffic in North American E&M signaling format to convert the signaling to MELCAS format, enter the following commands:
voice-port 1/0/0define rx-bits idle 1101define rx-bits seize 0101define tx-bits idle 1101define tx-bits seize 0101In this example, reverse polarity is configured on a voice port on a Cisco 3700 series router that is sending traffic in LMR signaling format:
voice-port 1/0/0define rx-bits idle 1111define rx-bits seize 0000define tx-bits idle 1111define tx-bits seize 0000Related Commands
Command Descriptioncondition
Manipulates the signaling bit-pattern for all voice signaling types.
ignore
Configures a North American E&M or E&M MELCAS voice port to ignore specific receive bits.
digital-filter
To specify the digital filter to be used before the voice packet is sent from the digital signal processor (DSP) to the network, use the digital-filter command in voice-class configuration mode. To remove the digital filter, use the no form of this command.
digital-filter {1950hz | 2175hz}
no digital-filter {1950hz | 2175hz}
Syntax Description
Defaults
Digital filtering is disabled.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
The digital-filter command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). The digital filter improves voice quality by preventing transmission of the guard tone with the voice packet from the LMR system to the VoIP network. The guard tone is configured with the inject guard-tone command. The digital filter can be configured to filter out either 2175 Hz or 1950 Hz. Only one of these frequencies can be filtered out at a time. Filtering is performed by the DSP.
Examples
The following example specifies that 1950 Hz guard tone be filtered out of the voice packet before it is sent from the DSP to the network:
voice class tone-signal mytonesdigital-filter 1950hzRelated Commands
ds0-group (E1)
To specify the DS0 time slots that make up a logical voice port on an E1 controller, specify the signaling type by which the router communicates with the PBX or PSTN, and define E1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN, use the ds0-group command in controller configuration mode. To remove the group and signaling setting, use the no form of this command.
Cisco 1750 and Cisco 1751
ds0-group ds0-group-number timeslots timeslot-list {[service service-type] | [type e&m-fgb | e&m-fgd | e&m-immediate-start | fgd-eana | fgd-os | fxs-ground-start | fxs-loop-start | none | r1-itu | r1-modified | r1-turkey | sas-ground-start | sas-loop-start]}
no ds0-group ds0-group-number
Cisco 2600 Series (Except Cisco 2691), Cisco 3600 Series (Except Cisco 3660)
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start | e&m-melcas-delay | e&m-melcas-immed | e&m-melcas-wink | e&m-wink-start | ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start | fxo-melcas | fxs-ground-start | fxs-loop-start | fxs-melcas | r2-analog | r2-digital | r2-pulse}
no ds0-group ds0-group-number
Cisco 2691, Cisco 2600XM Series, Cisco 2800 Series (Except Cisco 2801), Cisco 3660, Cisco 3700 Series, Cisco 3800 Series
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start | e&m-lmr |e&m-melcas-delay | e&m-melcas-immed | e&m-melcas-wink | e&m-wink-start | ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start | fxo-melcas | fxs-ground-start | fxs-loop-start | fxs-melcas | r2-analog | r2-digital | r2-pulse}
no ds0-group ds0-group-number
Cisco 7200 Series and Cisco 7500 Series Voice Ports
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-wink-start | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}
no ds0-group ds0-group-number
Cisco 7700 Series Voice Ports
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start | e&m-wink-start | fxs-ground-start | fxs-loop-start | fxo-ground-start | fxo-loop-start}
no ds0-group ds0-group-number
Cisco AS5300 and the Cisco AS5400
ds0-group ds0-group-number timeslots timeslot-list type {none | p7 | r2-analog | r2-digital | r2-lsv181-digital | r2-pulse}
no ds0-group ds0-group-number
Note
This command does not support the extended echo canceller (EC) feature on the Cisco AS5x00 series.
Syntax Description
Defaults
There is no DS0 group. Calls are allowed in both directions.
Command Modes
Controller configuration
Command History
Usage Guidelines
The ds0-group command automatically creates a logical voice port that is numbered as follows:
•
Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, and Cisco 3745, and Cisco 7200 series:
–
slot/port:ds0-group-number
Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
Be sure you take the following into account when you are configuring DS0 groups:
•
Channel groups, CAS voice groups, DS0 groups, and time-division multiplexing (TDM) groups all use group numbers. All group numbers configured for channel groups, CAS voice groups, DS0 groups, and TDM groups must be unique on the local router. For example, you cannot use the same group number for a channel group and for a TDM group.
•
The keywords available for the ds0-group command are dependent upon the Cisco IOS software release that you are using. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
http://www.cisco.com/go/fn•
When you are using command-line interface (CLI) help, the keywords for the ds0-group command are configuration specific. For example, if Media Gateway Control Protocol (MGCP) is configured, you see the mgcp keyword. If you are not using MGCP, you do not see the mgcp keyword.
Examples
The following example shows ranges of E1 controller time slots configured for FXS ground-start and FXO loop-start signaling:
E1 1/0framing esflinecode b8zsds0-group 1 timeslots 1-10 type fxs-ground-startds0-group 2 timeslots 11-24 type fxo-loop-startThe following example shows ranges of T1 controller time slots configured for FXS ground-start signaling:
controller E1 1/0ds0-group 1 timeslots 1-4 type fxs-ground-startThe following example illustrates setting the E1 channels for Signaling System 7 (SS7) service on any trunking gateway using the mgcp keyword:
Router(config-controller)# ds0-group 0 timeslots 1-24 type none service mgcpIn the following example, the time slot maximum is 12 and the time slot is 1, so two voice-ports are created successfully.
controller E1 0/0ds0-group 0 timeslots 1-4 type e&m-immediate-startds0-group 1 timeslots 6-12 type e&m-immediate-startIf a third DS0 group is added, the voice-port is rejected even though the total number of voice channels is less than 16.
ds0-group 2 timeslots 17-18 type e&m-immediate-startIn the following example, the signaling type is set to e&m-lmr:
ds0-group 0 timeslots 1-10 type e&m-lmrRelated Commands
ds0-group (T1)
To specify the DS0 time slots that make up a logical voice port on a T1 controller, to specify the signaling type by which the router communicates with the PBX or PSTN, and to define T1 channels for compressed voice calls and the channel-associated signaling (CAS) method by which the router connects to the PBX or PSTN, use the ds0-group command in controller configuration mode. To remove the group and signaling setting, use the no form of this command.
Cisco 1750 and Cisco 1751
ds0-group ds0-group-number timeslots timeslot-list [service service-type] type {e&m-fgb | e&m-fgd | e&m-immediate-start | fgd-eana | fgd-os | fxs-ground-start | fxs-loop-start | none | r1-itu | r1-modified | r1-turkey | sas-ground-start | sas-loop-start}
no ds0-group ds0-group-number
Cisco 2600 Series (Except Cisco 2691), Cisco 3600 Series (Except Cisco 3660), and Cisco VG 200
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-wink-start | ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start}
no ds0-group ds0-group-number
Cisco 2691, Cisco 2600XM Series, Cisco 2800 Series (Except Cisco 2801), Cisco 3660, Cisco 3700 Series, Cisco 3800 Series
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-lmr | e&m-wink-start | ext-sig | fgd-eana | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start}
no ds0-group ds0-group-number
Cisco 7200 Series and Cisco 7500 Series
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-fgd | e&m-immediate-start | e&m-wink-start | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start}
no ds0-group ds0-group-number
Cisco 7700 Series Voice Ports
ds0-group ds0-group-number timeslots timeslot-list type {e&m-delay-dial | e&m-immediate-start | e&m-wink-start | fxo-ground-start | fxo-loop-start | fxs-ground-start | fxs-loop-start}
no ds0-group ds0-group-number
Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850
ds0-group ds0-group-number timeslots timeslot-list [service service-type] [type [e&m-fgb [dtmf | mf] | e&m-fgd [dtmf | mf [dnis | ani-dnis [info-digits-no-strip] | service service-type] | e&m-immediate-start | fxs-ground-start | fxs-loop-start | fgd-eana [ani-dnis | mf] | fgd-os [dnis-ani | mf] | r1-itu [dnis] | sas-ground-start | sas-loop-start | none]]
no ds0-group ds0-group-number
Syntax Description
ds0-group-number
A value that identifies the DS0 group. Range is from 0 to 23.
timeslots timeslot-list
Lists time slots in the DS0 group. The timeslot-list argument is a single time-slot number, a single range of numbers, or multiple ranges of numbers separated by commas. Range is from 1 to 24. Examples are as follows:
•
2
•
1-15,17-24
•
1-23
•
2,4,6-12
type
Specifies the type of signaling for the DS0 group. The signaling method selection for the type keyword depends on the connection that you are making. The ear and mouth (E&M) interface allows connection for PBX trunk lines (tie lines) and telephone equipment. The Foreign Exchange Station (FXS) interface allows connection of basic telephone equipment and PBX. The Foreign Exchange Office (FXO) interface is for connecting the central office (CO) to a standard PBX interface where permitted by local regulations; it is often used for off-premise extensions (OPXs). Types are as follows:
•
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-fgb—E&M Type II Feature Group B.
•
e&m-fgd—E&M Type II Feature Group D.
•
e&m-immediate-start—E&M immediate start.
•
e&m-lmr—E&M Land Mobile Radio (LMR).
•
e&m-wink-start—The originating endpoint sends an off-hook signal and waits for a wink-start from the destination.
•
ext-sig—The external signaling interface specifies that the signaling traffic comes from an outside source.
•
fgd-eana—Feature Group D exchange access North American.
•
fgd-os—Feature Group D operator services.
•
fxo-ground-start—FXO ground-start signaling.
•
fxo-loop-start—FXO loop-start signaling.
•
fxs-ground-start—FXS ground-start signaling.
•
fxs-loop-start—FXS loop-start signaling.
•
none—Null signaling for external call control.
•
r1-itu—Line signaling based on international signaling standards.
•
r1-modified—An international signaling standard that is common to channelized T1/E1 networks.
•
r1-turkey—A signaling standard used in Turkey.
•
sas-ground-start—Single attachment station (SAS) ground-start.
•
sas-loop-start—SAS loop-start.
service service-type
(Optional) Specifies the type of service.
•
data—Data service.
•
fax— Store-and-forward fax service.
•
mgcp1 —Media Gateway Control Protocol service.
•
sccp1—Simple Gateway Control Protocol service
•
voice—Voice service (for FGD-OS service).
dtmf
(Optional) Specifies dual tone multifrequency (DTMF) tone signaling.
mf
(Optional) Specifies multifrequency (MF) tone signaling
ani-dnis
(Optional) Specifies automatic number identification (ANI) and dialed number identification service (DNIS) address information provisioning for FGD OS.
dnis-ani
(Optional) Specifies ANI and DNIS address information provisioning for FGD EANA.
dnis
(Optional) Specifies DNIS address information provisioning.
info-digits-no-strip
(Optional) Retains info digits on the Cisco AS5x00 platforms.
1 Used only with the type none keywords on the Cisco AS5x00 platforms.
Defaults
There is no DS0 group. Calls are allowed in both directions.
Command Modes
Controller configuration
Command History
Usage Guidelines
The ds0-group command automatically creates a logical voice port that is numbered as follows:
•
Cisco 2600 series, Cisco 2600XM, Cisco 3660, Cisco 3725, Cisco 3745, and Cisco 7200 series:
–
slot/port:ds0-group-number
•
Cisco AS5300 with a T1 controller:
–
slot/port
•
Cisco AS5850 with a T1 controller:
–
slot/port:ds0-group-number
Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
Be sure that you take the following into account when you are configuring DS0 groups:
•
Channel groups, CAS voice groups, DS0 groups, and time-division multiplexing (TDM) groups all use group numbers. All group numbers configured for channel groups, CAS voice groups, DS0 groups, and TDM groups must be unique on the local router. For example, you cannot use the same group number for a channel group and for a TDM group.
•
The keywords available for the ds0-group command are dependent upon the Cisco IOS software release that you are using. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
•
When you are using command-line interface (CLI) help, the keywords for the ds0-group command are configuration specific. For example, if Media Gateway Control Protocol (MGCP) is configured, you see the mgcp keyword. If you are not using MGCP, you do not see the mgcp keyword.
Note
This command does not support the extended echo canceller (EC) feature on the Cisco AS5x00 series.
Examples
The following example shows ranges of T1 controller time slots configured for FXS ground-start and FXO loop-start signaling:
controller T1 1/0framing esflinecode b8zsds0-group 1 timeslots 1-10 type fxs-ground-startds0-group 2 timeslots 11-24 type fxo-loop-startThe following example shows ranges of T1 controller time slots configured for FXS ground-start signaling:
controller T1 1/0ds0-group 1 timeslots 1-4 type fxs-ground-startThe following example illustrates setting the T1 channels for Signaling System 7 (SS7) service on any trunking gateway using the mgcp keyword:
ds0-group 0 timeslots 1-24 type none service mgcpIn the following example, the time slot maximum is 12 and the time slot is 1, so two voice-ports are created successfully.
controller T1 0/0ds0-group 0 timeslots 1-4 type e&m-immediate-startds0-group 1 timeslots 6-12 type e&m-immediate-startIf a third DS0 group is added, the voice port is rejected even though the total number of voice channels is less than 16.
ds0-group 2 timeslots 17-18 type e&m-immediate-startIn the following example, the signaling type is set to E&M LMR:
ds0-group 0 timeslots 1-10 type e&m-lmrYou have the option to retain info digits when you are configuring E&M Type II Feature Group D with MF signaling and ANI/DNIS for calls being sent over IP. Info digits denote the subscriber type, and the info-digits keyword prepends info digits to the calling number.
On inbound calls from a T1 FGD voice-port with MF ANI-DNIS, when ANI information is obtained, it is passed unaltered to the next matching dial peer, either POTS or VoIP. The addition of the info-digits-no-strip keyword allows you to retain the info digits portion of the ANI information; the modified ANI is then passed to the next matching dial peer. Ordinarily, info digits are not valid for calls going over IP and are, therefore, stripped off. The ability to retain info digits is particularly useful for calls that are not leaving the PSTN network and are just being hairpinned back.
In the following example, the E&M Type II Feature Group D is configured with MF signaling and ANI/DNIS over IP while retaining info digits:
ds0-group 0 timeslots 1-24 type e&m-fgd mf ani-dnis info-digits-no-stripRelated Commands
inject guard-tone
To play out a guard tone with the voice packet, use the inject guard-tone command in voice-class configuration mode. To remove the guard tone, use the no form of this command.
inject guard-tone frequency amplitude [idle]
no inject guard-tone frequency amplitude [idle]
Syntax Description
Defaults
No guard tone is injected.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
The inject guard-tone command has an effect on an ear and mouth (E&M) analog or digital voice port only if the signal type for that port is Land Mobile Radio (LMR). The guard tone is played out with the voice packet to keep the radio channel up. Guard tones of 1950 Hz and 2175 Hz can be filtered out before the voice packet is sent from the digital signal processor (DSP) to the network using the digital-filter command.
Examples
The following example configures a guard tone of 1950 Hz and -10 dBm to be played out with voice packets:
voice class tone-signal tone1inject guard-tone 2175 -30Related Commands
Command Descriptiondigital-filter
Specifies the digital filter to be used before the voice packet is sent from the DSP to the network.
inject pause
To specify a pause between injected tones, use the inject pause command in voice-class configuration mode. To remove the pause, use the no form of this command.
inject pause index milliseconds
no inject pause index milliseconds
Syntax Description
index
Order of pauses and tones. Range is integers from 1 to 10.
milliseconds
Duration, in milliseconds, of the pause between injected tones. Range is integers from 10 to 500.
Defaults
milliseconds: 0 milliseconds
Command Modes
Voice-class configuration
Command History
Usage Guidelines
The inject pause command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). Use this command to specify the pause between injected tones specified with the inject tone command. Use the index argument of this command in conjunction with the index argument of the inject tone command to specify the order of the pauses and tones.
Examples
The following example configures a pause of 100 milliseconds after the injected tone:
voice class tone-signal 100inject tone 1 2000 0 200inject pause 2 100Related Commands
Command Descriptioninject tone
Specifies a wakeup or frequency selection tone to be played out before the voice packet.
inject tone
To specify a wakeup or frequency selection tone to be played out before the voice packet, use the inject tone command in voice-class configuration mode. To remove the tone, use the no form of this command.
inject tone index frequency amplitude duration
no inject tone index frequency amplitude duration
Syntax Description
Defaults
No tone is injected.
Command Modes
Voice-class configuration
Command History
Usage Guidelines
The inject tone command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). Use this command with the inject pause command to configure wakeup and frequency selection tones. Use the index argument of this command in conjunction with the index argument of the inject pause command to specify the order of the pauses and tones.
If you configure injected tones with this command, be sure to use the timing delay-voice tdm command to configure a delay before the voice packet is played out. The delay must be equal to the sum of the durations of the injected tones and pauses in the tone-signal voice class.
Examples
The following example configures a frequency selection tone to be played out before the voice packet:
voice class tone-signal 100inject tone 1 1950 3 150inject tone 2 2000 0 60inject pause 3 60inject tone 4 2175 3 150inject tone 5 1000 0 50Related Commands
Command Descriptioninject pause
Specifies a pause between injected tones.
timing delay-voice tdm
Specifies the delay before a voice packet is played out.
input gain
To configure a specific input gain value or enable automatic gain control, use the input gain command in voice-port configuration mode. To disable the selected amount of inserted gain, use the no form of this command.
input gain {decibels | auto-control [auto-dbm]}
no input gain {decibels | auto-control [auto-dbm]}
Syntax Description
Defaults
decibels: 0 decibels
auto-dbm: -9 dBmCommand Modes
Voice-port configuration
Command History
Usage Guidelines
A system-wide loss plan must be implemented using both the input gain and output attenuation commands. You must consider other equipment (including PBXs) in the system when creating a loss plan. The default value for this command assumes that a standard transmission loss plan is in effect, meaning that there is typically a minimum attenuation of -6 dB between phones, especially if echo cancellers are present. Connections are implemented to provide 0 dB of attenuation when the input gain and output attenuation commands are configured with the default value of 0 dB.
You cannot increase the gain of a signal to the public switched telephone network (PSTN), but you can decrease it. If the voice level is too high, you can decrease the volume by either decreasing the input gain or increasing the output attenuation.
You can increase the gain of a signal coming into the router. If the voice level is too low, you can increase the input gain by using the input gain command.
Typical Land Mobile Radio (LMR) signaling systems send 0 dB out and expect -10 dB in. Setting output attenuation to 10 dB is typical. Output attenuation should be adjusted to provide the voice level required by the radio to produce correct transmitter modulation.
The auto-control keyword and auto-dbm argument are available on an ear and mouth (E&M) voice port only if the signal type for that port is LMR. The auto-control keyword enables automatic gain control, which is performed by the digital signal processor (DSP). Automatic gain control adjusts speech to a comfortable volume when it becomes too loud or too soft. Because of radio network loss and other environmental factors, the speech level arriving at a router from an LMR system could be very low. You can use automatic gain control to ensure that the speech is played back at a more comfortable level. Because the gain is inserted digitally, the background noise can also be amplified. Automatic gain control is implemented as follows:
•
Output level: -9 dB
•
Gain range: -12 dB to 20 dB
•
Attack time (low to high): 30 milliseconds
•
Attack time (high to low): 8 seconds
Examples
The following example inserts a 3-dB gain at the receiver side of the interface in the Cisco 3600 series router:
port 1/0/0input gain 3Related Commands
Command Descriptionoutput attenuation
Configures a specific output attenuation value or enables automatic gain control for a voice port.
lmr duplex half
To have the voice path for a voice port operate in half duplex mode, use the lmr duplex half command in voice-port configuration mode. To return to the default, use the no form of this command.
lmr duplex half
no lmr duplex half
Syntax Description
This command has no arguments or keywords.
Defaults
Full duplex mode
Command Modes
Voice-port configuration
Command History
Usage Guidelines
When a radio system is receiving voice traffic from the radio, operating the voice path in half duplex mode prevents the speaker from being interrupted and prevents the voice stream from being fed back to itself.
Examples
In the following example, the voice path for voice port 1/0/0 on a Cisco 3700 series router is set to operate in half duplex mode:
voice-port 1/0/0lmr duplex half
lmr e-lead
To define the use of the E-lead in signaling between the ear and mouth (E&M) voice port on the router and the attached Land Mobile Radio (LMR) device, use the lmr e-lead command in voice-port configuration mode. To return to the default use of the E-lead, use the no form of this command.
lmr e-lead {inactive | seize | voice}
no lmr e-lead {inactive | seize | voice}
Syntax Description
Defaults
seize
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The lmr e-lead command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is LMR. The lmr e-lead command is effective only if the attached LMR device operates under E-lead control. Use the lmr e-lead command to configure the voice port when using private line, automatic ringdown (PLAR) connections. The E-lead connects to the Push To Talk (PTT) of the LMR system.
Examples
In the following example, packet transmission from the E&M voice port on a Cisco 3745 to an attached LMR radio system is disabled:
lmr e-lead inactiveRelated Commands
Command Descriptionlmr m-lead
Defines the use of the M-lead in signaling between the E&M voice port on the router and the attached LMR device.
lmr led-on
To use the ear and mouth (E&M) LED to indicate the E-lead and M-lead status, use the lmr led-on command in voice-port configuration mode. To return to the default use of the E&M LED, use the no form of this command.
lmr led-on
no lmr led-on
Syntax Description
This command has no arguments or keywords.
Defaults
The E&M LED indicates voice port activity only.
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The lmr e-lead command is available on an E&M voice port only if the signal type for that port is Land Mobile Radio (LMR). This command enables the use of the E&M LED to indicate the E-lead and M-lead status as follows:
•
Red—E-lead active
•
Green—M-lead active
•
Yellow—Both E-lead and M-lead active
The default behavior of the E&M LED is to light up when there is activity on the voice port and to turn off when there is no activity.
Examples
The following example specifies that the E&M LED is used to indicate the E-lead and M-lead status:
voice-port 1/0/0lmr led-on
lmr m-lead
To define the use of the M-lead in signaling between the ear and mouth (E&M) voice port on the router and the attached Land Mobile Radio (LMR) device, use the lmr m-lead command in voice-port configuration mode. To return to the default use of the M-lead, use the no form of this command.
lmr m-lead {inactive | audio-gate-in | dialin}
no lmr m-lead {inactive | audio-gate-in | dialin}
Syntax Description
Defaults
inactive
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The lmr m-lead command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is LMR. The lmr e-lead command is effective only if the attached LMR device operates under M-lead control. The M-lead corresponds to the Carrier Operated Relay (COR) of the LMR system, which indicates receive activity on the LMR system.
Examples
In the following example, an LMR radio system attached to the E&M voice port on a Cisco 3745 is allowed to transmit audio by first raising the E-lead, then transmitting:
lmr m-lead dialinRelated Commands
Command Descriptionlmr e-lead
Defines the use of the E-lead in signaling between the E&M voice port on the router and the attached LMR device.
music-threshold
To specify the threshold for on-hold music for a specified voice port, use the music-threshold command in voice-port configuration mode. To disable this feature, use the no form of this command.
music-threshold decibels
no music-threshold decibels
Syntax Description
decibels
On-hold music threshold, in decibels (dB). Range is from -70 to -10 (integers only). The default is -38 dB.
Defaults
-38 dB
Command Modes
Voice-port configuration
Command History
Usage Guidelines
Use this command to specify the decibel level of music played when calls are put on hold. This command tells the firmware to pass steady data above the specified level. It affects the operation of voice activity detection (VAD) only when the voice port is receiving voice.
If the value for this command is set too high, VAD interprets music-on-hold as silence, and the remote end does not hear the music. If the value for this command is set too low, VAD compresses and passes silence when the background is noisy, creating unnecessary voice traffic.
Examples
The following example sets the decibel threshold to -35 for the music played when calls are put on hold:
voice port 0:Dmusic-threshold -35The following example sets the decibel threshold to -35 for the music played when calls are put on hold on a Cisco 3600 series router:
voice-port 1/0/0music-threshold -35
output attenuation
To configure a specific output attenuation value or enable automatic gain control, use the output attenuation command in voice-port configuration mode. To disable the selected output attenuation value, use the no form of this command.
output attenuation {decibels | auto-control [auto-dbm]}
no output attenuation {decibels | auto-control [auto-dbm]}
Syntax Description
Defaults
For Foreign Exchange Office (FXO), Foreign Exchange Station (FXS), and ear and mouth (E&M)
ports: decibels: 0 decibels
auto-dbm: -9 dBmCommand Modes
Voice-port configuration
Command History
Usage Guidelines
A system-wide loss plan must be implemented using both the input gain and output attenuation commands. You must consider other equipment (including PBXs) in the system when creating a loss plan. The default value for this command assumes that a standard transmission loss plan is in effect, meaning that there must be an attenuation of -6 dB between phones. Connections are implemented to provide -6 dB of attenuation when the input gain and output attenuation commands are configured with the default value of 0 dB.
You cannot increase the gain of a signal to the public switched telephone network (PSTN), but you can decrease it. If the voice level is too high, you can decrease the volume by either decreasing the input gain or increasing the output attenuation.
You can increase the gain of a signal coming into the router. If the voice level is too low, you can increase the input gain by using the input gain command.
The auto-control keyword and auto-dbm argument are available on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). The auto-control keyword enables automatic gain control, which is performed by the digital signal processor (DSP). Automatic gain control adjusts speech to a comfortable volume when it becomes too loud or too soft. Because of radio network loss and other environmental factors, the speech level arriving at a router from an LMR system could be very low. You can use automatic gain control to ensure that the speech is played back at a more comfortable level. Because the gain is inserted digitally, the background noise can also be amplified. Automatic gain control is implemented as follows:
•
Output level: -9 dB
•
Gain range: -12 dB to 20 dB
•
Attack time (low to high): 30 milliseconds
•
Attack time (high to low): 8 seconds
Examples
On the Cisco 3600 series router, the following example configures a 3-dB loss to be inserted at the transmit side of the interface:
voice-port 1/0/0output attenuation 3On the Cisco 3600 series router, the following example configures a 3-dB gain to be inserted at the transmit side of the interface:
voice-port 1/0/0output attenuation -3On the Cisco AS5300, the following example configures a 3-dB loss to be inserted at the transmit side of the interface:
voice-port 0:Doutput attenuation 3Related Commands
show voice lmr
To display the Land Mobile Radio (LMR) related dynamic information and static information for LMR ports or a ds0 group, use the show voice lmr command in EXEC mode.
show voice lmr [slot/subunit/port | slot/port:ds0-group] [details]
Syntax Description
Command Modes
EXEC
Command History
Usage Guidelines
This command displays information for LMR voice ports only. If no voice port is specified, the command displays information for all E&M LMR voice ports.
When the details keyword is used, this command displays information about timeouts, timers, and injected tones and pauses, in addition to detailed voice port and active call information found in the show voice port and show call active voice commands.
Examples
The following is sample output from the show voice lmr command for an E&M LMR analog voice port on a Cisco 3745 router:
Router# show voice lmr 2/0/02/0/0=========Connection type: n/aOut Attenuation = 0 db, In Gain = 0 dBE-lead capability is inactive, polarity = normalM-lead capability is inactive, polarity = normalvoice-class tone-signal teststate = LMR_CONNECT, e-lead = off, m-lead = offfull duplex, voice path = rxTerminating side of the connectionTransmitPackets=113, TransmitBytes=2241ReceivePackets=113, ReceiveBytes=2241CoderTypeRate=g729r8NoiseLevel=-65, ACOMLevel=22OutSignalLevel=-68, InSignalLevel=-79RemoteIPAddress=10.5.25.40, RemoteUDPPort=17272Remote SignallingIPAddress=10.5.25.40, Port=15418Remote MediaIPAddress=10.5.25.40, Port=17272RoundTripDelay=2 msSessionProtocol=ciscoVAD =enabledThe following is sample output from the show voice lmr details command for an E&M LMR analog voice port on a Cisco 3745 router:
Router# show voice lmr 2/0/0 details2/0/0=========Description:Connection type: n/aOut Attenuation = 0 db, In Gain = 0 dBTiming hangover: 500 msE-lead capability is inactive, polarity = normalM-lead capability is inactive, polarity = normalTiming hookflash-in: 480Timing delay-voice: 470 msMusic On Hold Threshold: -38 dB, Noise Threshold: -62 dBE&M type: 1, Operation: 2-wireImpedance is set to 600r Ohmlmr tear down timeout is set to 1800 secondlmr PTT transmit timeout is not setlmr PTT receive timeout is not setvoice-class tone-signal testinject tone 1 1950 3 150inject tone 2 2000 0 60inject pause 3 60inject tone 4 2175 3 150inject tone 5 1000 0 50inject guard-tone 6 1950 -10state = LMR_CONNECT, e-lead = off, m-lead = offfull duplex, voice path = rxTerminating side of the connectionTransmitPackets=113, TransmitBytes=2241ReceivePackets=113, ReceiveBytes=2241CoderTypeRate=g729r8NoiseLevel=-66, ACOMLevel=22OutSignalLevel=-68, InSignalLevel=-79PeerAddress=37200PeerSubAddress=PeerId=200SessionTarget=RemoteIPAddress=10.5.25.40, RemoteUDPPort=17272Remote SignallingIPAddress=10.5.25.40, Port=15418Remote MediaIPAddress=10.5.25.40, Port=17272RoundTripDelay=0 msSessionProtocol=ciscoVAD =enabledSelectedQoS=best-effortProtocolCallId=SessionTarget=Table 6 describes the significant fields shown in the output, in the order in which they appear.
Related Commands
Command Descriptionshow call active voice
Displays call information for voice calls in progress.
show voice port
Displays configuration information about a specific voice port.
rtp payload-type
To identify the payload type of a Real-Time Transport Protocol (RTP) packet, use the rtp payload-type command in dial peer voice configuration mode. To remove the RTP payload type, use the no form of this command.
rtp payload-type {cisco-cas-payload number | cisco-clear-channel number | cisco-codec-fax-ack number | cisco-codec-fax-ind number | cisco-codec-gsmamrnb number | cisco-codec-ilbc number | cisco-codec-video-h263+ number | cisco-codec-video-h264 number | cisco-fax-relay number | cisco-pcm-switch-over-alaw number | cisco-pcm-switch-over-ulaw number | cisco-rtp-dtmf-relay number | lmr-tone number | nse number | nte number | nte-tone number} [comfort-noise {13 | 19}]
no rtp payload-type {cisco-cas-payload number | cisco-clear-channel number | cisco-codec-fax-ack number | cisco-codec-fax-ind number | cisco-codec-gsmamrnb number | cisco-codec-ilbc number | cisco-codec-video-h263+ number | cisco-codec-video-h264 number | cisco-fax-relay number | cisco-pcm-switch-over-alaw number | cisco-pcm-switch-over-ulaw number | cisco-rtp-dtmf-relay number | lmr-tone number | nse number | nte number | nte-tone number} [comfort-noise {13 | 19}]
Syntax Description
Command Default
No RTP payload type is configured.
Command Modes
Dial peer voice configuration
Command History
Usage Guidelines
Use this command to identify the payload type of an RTP. Use this command after the dtmf-relay command is used to choose the NTE method of DTMF relay for a Session Initiation Protocol (SIP) call.
Configured payload types of NSE and NTE exclude certain values that have been previously hard-coded with Cisco-proprietary meanings. Do not use the following numbers, which have preassigned values: 96, 97, 100, 117, 121 to 123, and 125 to 127.
Use of any of these values results in an error message when the command is entered. You must first reassign the value in use to a different unassigned number; for example:
rtp payload-type cisco-codec-ilbc 100ERROR: value 100 in use!rtp payload-type nse 105rtp payload-type cisco-codec-ilbc 100Examples
The following example identifies the RTP payload type as GSMAMR-NB124:
Router(config-dial-peer)# rtp payload-type cisco-codec-gsmamrnb 124The following example identifies the RTP payload type as NTE 99:
Router(config-dial-peer)# rtp payload-type nte 99The following example identifies the RTP payload type for the iLBC as 100:
Router(config-dial-peer)# rtp payload-type cisco-codec-ilbc 100Related Commands
Command Descriptiondtmf-relay
Specifies how an H.323 or SIP gateway relays DTMF tones between telephony interfaces and an IP network.
show voip rtp connections
To display Real-Time Transport Protocol (RTP) named event packets, use the show voip rtp connections command in privileged EXEC mode.
show voip rtp connections [detail]
Syntax Description
Command Modes
Privileged EXEC
Command History
Usage Guidelines
This command displays information about RTP named event packets, such as caller ID number, IP address, and port for both the local and remote endpoints. The output from this command provides an overview of all the connections in the system, and this information can be used to narrow the criteria for debugging. The debug voip rtp command floods the console with voice packet information. You can use the show voip rtp connections command to get caller ID, remote IP address, or remote port identifiers that you can use to limit the output from the debug voip rtp command.
The detail keyword allows you to identify the phone or phones that have connected two RTP call legs together to create VoIP-to-VoIP or VoIP-to-POTS hairpins. If the detail keyword is omitted, the output does not display calls that are connected by hairpin call routing.
Examples
Table 7 describes the significant fields shown in the examples. Each line of output under "VoIP RTP active connections" shows information for one call leg. A phone call normally consists of two call legs, one connected to the calling party and one connected to the called party. The router joins (or bridges) the two call legs together to make a call. The show voip rtp connections command shows the RTP information for H.323 and SIP calls only; it does not directly show the POTS call legs. The information for the IP phone can be seen using the show ephone offhook command.
The following sample output shows an incoming H.323 call that is being directed to an IP phone attached to a Cisco CME system.
Router# show voip rtp connectionsVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 21 22 16996 18174 10.4.204.37 10.4.204.24Found 1 active RTP connectionsThe following sample output shows the same call as in the previous example, but using the detail keyword with the command. The sample output shows the called number (1509) and calling number (8108) on both call legs (21 and 22); the called and calling numbers are the same on both legs for a simple A-to-B call. Leg 21 is the H.323 segment of the and leg 22 is the POTS segment that goes to the IP phone.
Router# show voip rtp connections detailVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 21 22 16996 18174 10.4.204.37 10.4.204.24callId 21 (dir=1):called=1509 calling=8108 redirect=dest callId 22:called=1509 calling=8108 redirect=1 context 64FB3358 xmitFunc 6032E8B4Found 1 active RTP connectionsThe following example shows the call from the previous example being transferred by extension 1509 to extension 1514. Notice that the dstCallId changed from 22 to 24, but the original call leg (21) for the transferred party is still present. This implies that H.450.2 capability was disabled for this particular call, because if H.450.2 was being used for the transfer, the transfer would have caused the incoming H.323 call leg to be replaced with a new call.
Router# show voip rtp connectionsVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 21 24 16996 18174 10.4.204.37 10.4.204.24Found 1 active RTP connectionsThe following example shows the detailed output for the same transfer as shown in the previous example. The original incoming call leg is still present (21) and still has the original called and calling numbers. The transferred call leg (24) shows 1509 (the transferring party) as the calling party and 1514 (the transfer destination) as the called party.
Router# show voip rtp connections detailVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 21 24 16996 18174 10.4.204.37 10.4.204.24callId 21 (dir=1):called=1509 calling=8108 redirect=dest callId 24:called=1514 calling=1509 redirect=1 context 6466E810 xmitFunc 6032E8B4Found 1 active RTP connectionsThe following sample output shows a cross-linked call with two H.323 call legs. The first line of output shows that the CallID for the first call leg is 7 and that this call leg is associated with another call leg that has a destination CallId of 8. The next line shows that the CallID for the leg is 8 and that it is associated with another call leg that has a destination CallId of 7. This cross-linkage between CallIds 7 and 8 shows that the first call leg is related to the second call leg (and vice versa). From this you can infer that the two call legs are actually part of the same phone call.
In an active system you can expect many lines of output that you would have to sort through to see which ones have this cross-linkage relationship. The lines showing two related call legs are not necessarily listed in adjacent order.
Router# show voip rtp connectionsVoIP RTP active connections :No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP1 7 8 16586 22346 172.27.82.2 172.29.82.22 8 7 17010 16590 172.27.82.2 192.168.1.29Found 2 active RTP connections
Related Commands
Command Descriptiondebug voip rtp
Enables debugging for RTP named event packets.
show ephone offhook
Displays information and packet counts for phones that are currently off hook.
signal
To specify the type of signaling for a voice port, use the signal command in voice-port configuration mode. To reset to the default, use the no form of this command.
Foreign Exchange Office (FXO) and Foreign Exchange Station (FXS) Voice Ports
signal {loop-start | ground-start}
no signal {loop-start | ground-start}
Ear and mouth (E&M) Voice Ports
signal {wink-start | immediate | delay-dial | lmr}
no signal {wink-start | immediate | delay-dial | lmr}
Centralized Automatic Message Accounting (CAMA) Ports
signal {cama {kp-0-nxx-xxxx-st | kp-0-npa-nxx-xxxx-st | kp-2-st | kp-npd-nxx-xxxx-st}
| groundstart | loopstart}no signal {cama {kp-0-nxx-xxxx-st | kp-0-npa-nxx-xxxx-st | kp-2-st | kp-npd-nxx-xxxx-st} | groundstart | loopstart}
Syntax Description
Defaults
FXO and FXS interfaces: loop-start
E&M interfaces: wink-start
CAMA interfaces: loop-startCommand Modes
Voice-port configuration
Command History
Usage Guidelines
This command applies to analog voice ports only.
Using the signal command for an FXO or FXS voice port changes the signal value for both voice ports on a voice port module (VPM) card.
Note
If you change the signal type for an FXO voice port on Cisco 3600 series routers, you need to move the appropriate jumper in the voice interface card of the voice network module. For more information about the physical characteristics of the voice network module, refer to the installation documentation, Voice Network Module and Voice Interface Card Configuration Note, that came with your voice network module.
Configuring this command for an E&M voice port changes only the signal value for the selected voice port. In either case, the voice port must be shut down and then activated before the configured values take effect.
Some PBXs miss initial digits if the E&M voice port is configured for immediate start signaling. Immediate start signaling should be used for dial pulse outpulsing only and only on circuits for which the far end is configured to accept digits within a few milliseconds of seizure. Delay dial signaling, which is intended for use on trunks and not lines, relies on the far end to return an off-hook indication on its M-lead as soon as the circuit is seized. When a receiver is attached, the far end removes the off-hook indication to indicate that it is ready to receive digits. Delay dial must be configured on both ends to work properly. Some non-Cisco devices have a limited number of DTMF receivers. This type of equipment must delay the calling side until a DTMF receiver is available.
To specify which VIC-2CAMA ports are designated as dedicated CAMA ports for emergency 911 calls, use the signal cama command. No two service areas in the existing North American telephony infrastructure supporting E911 calls have identical service implementations, and many of the factors that drive the design of emergency call handling are matters of local policy and therefore outside the scope of this document. Local policy determines which ANI format is appropriate for the specified Physical Service Access Point (PSAP) location.
The following four types of ANI transmittal schemes are based on the actual number of digits transmitted toward the E911 tandem. In each instance, the actual calling number is proceeded with a key pulse (KP) followed by an information (I) field or a NPD, which is then followed by the ANI calling number, and finally is followed by a start pulse (ST), STP, ST2P, or ST3P, depending on the trunk group type in the PSTN and the traffic mix carried.
The information field is one or two digits, depending on how the circuit was ordered originally. For one-digit information fields, a value of 0 indicates that the calling number is available. A value of 1 indicates that the calling number is not available. A value of 2 indicates an ANI failure. For a complete list of values for two-digit information fields, refer to SR-2275: Telcordia Notes on the Networks at www.telcordia.com.
•
7-digit transmission (kp-0-nxx-xxxx-st):
The calling phone number is transmitted, and the NPA is implied by the trunk group and not transmitted.
•
8-digit transmission (KP-npd-nxx-xxxx-st):
The I field consists of single-digit NPD-to-NPA mapping. When the calling party number of 415-555-0122 places a 911 call, and the Cisco 2600 series or Cisco 3600 series has an NPD (0)-to-NPA (415) mapping, the NPA signaling format is received by the selective router at the central office (CO).
Note
NPD values greater than 3 are reserved for signifying error conditions.
•
10-digit transmission (kp-0-npa-nxx-xxxx-st):
The E.164 number is fully transmitted.
•
kp-2-st transmission (kp-2-st):
kp-2-st transmission is used if the PBX is unable to out-pulse the ANI. If the ANI received by the Cisco router is not as per configured values, kp-2-st is transmitted. For example, if the voice port is configured for out-pulsing a ten-digit ANI and the 911 call it receives has a seven-digit calling party number, the router transmits kp-2-st.
Note
Emergency 911 calls are not rejected for an ANI mismatch. The call establishes a voice path. The E911 network, however, does not receive the ANI.
Examples
The following example configures ground-start signaling on the Cisco 3600 series as the signaling type for a voice port, which means that both sides of a connection can place a call and hang up:
voice-port 1/1/1signal ground-startThe following example configures a ten-digit ANI transmission:
Router(config)# voice-port 1/0/0Router(config-voiceport)# signal cama kp-0-npa-nxx-xxxx-stRelated Commands
signal keepalive
To configure the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks, use the signal keepalive command in voice-class configuration mode. To reset to the default, use the no form of this command.
signal keepalive {seconds | disabled}
no signal keepalive {seconds | disabled}
Syntax Description
seconds
Keepalive signaling packet interval, in seconds. Range is from 1 to 65535. Default is 5 seconds.
disabled
Specifies that no keepalive signals are sent.
Defaults
seconds: 5 seconds
Command Modes
Voice-class configuration
Command History
Usage Guidelines
Before configuring the keepalive signaling interval, you must use the voice class permanent command in global configuration mode to create a voice class for the Cisco trunk or FRF.11 trunk. The voice class must then be assigned to a dial peer using the voice-class permanent (dial-peer) command.
To avoid sending keepalive signals to a multicasting network with no specified destination, we recommend that you use the disabled keyword when configuring this command for use in networks that use connection trunk connections and multicasting.
Examples
The following example shows the keepalive signaling interval set to 3 seconds for voice class 10:
voice class permanent 10signal keepalive 3exitdial-peer voice 100 vofrvoice-class permanent 10Related Commands
test lmr clear-call
To tear down a Land Mobile Radio (LMR) connection, use the test lmr clear-call command in privileged EXEC mode.
test lmr {slot/subunit/port | slot/port:ds0-group} clear-call
Syntax Description
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Because the LMR signaling protocol cannot terminate a call, the test lmr clear-call command can be used to tear down the call manually. This command tears down all LMR connections on the specified voice port.
Examples
In this example, all existing LMR connections on voice port 1/0/0 are torn down:
test lmr 1/0/0 clear-calltest lmr clear-call
To tear down a Land Mobile Radio (LMR) connection, use the test lmr clear-call command in privileged EXEC mode.
test lmr {slot/subunit/port | slot/port:ds0-group} clear-call
Syntax Description
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Because the LMR signaling protocol cannot terminate a call, the test lmr clear-call command can be used to tear down the call manually. This command tears down all LMR connections on the specified voice port.
Examples
In this example, all existing LMR connections on voice port 1/0/0 are torn down:
test lmr 1/0/0 clear-call
timeout ptt
To specify a maximum time for transmitting or receiving a voice packet, use the timeout ptt command in voice-port configuration mode. To return to the default, use the no form of this command.
timeout ptt {rcv | xmt} minutes
no timeout ptt {rcv | xmt} minutes
Syntax Description
Defaults
minutes: 0 minutes
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The timeout ptt command is available on an ear and mouth (E&M) analog or digital voice port only if the signal type for that port is Land Mobile Radio (LMR). The purpose of this command is to limit extended radio transmission. When the time limit configured with this command expires, the radio transmitter unkeys, so that listeners on the channel cannot hear the speaker, even if the speaker continues to talk. When the speaker unkeys the radio, the timer is reactivated.
Examples
The following example specifies a maximum time of 10 minutes for transmitting a voice packet:
voice-port 1/0/0timeout ptt xmt 10
timeouts teardown lmr
To configure the time for which a Land Mobile Radio (LMR) voice port waits before tearing down an LMR connection after detecting no voice activity, use the timeouts teardown lmr command in voice-port configuration mode. To reset to the default, use the no form of this command.
timeouts teardown lmr {seconds | infinity}
no timeouts teardown lmr {seconds | infinity}
Syntax Description
Defaults
180 seconds
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The timeouts teardown lmr command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is LMR.
Examples
The following example configures voice port 1/0/1 on a Cisco 3745 to remain connected for 6 seconds after no voice activity is detected by the voice port:
voice-port 1/0/1timeouts teardown lmr 6Related Commands
timing delay-voice tdm
To specify the delay after which voice packets are played out, use the timing delay-voice tdm command in voice-port configuration mode. To reset to the default, use the no form of this command.
timing delay-voice tdm milliseconds
no timing delay-voice tdm milliseconds
Syntax Description
milliseconds
Duration, in milliseconds, of the timing delay. Range is integers from 1 to 1500. Default is 0.
Defaults
milliseconds: 0 milliseconds
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The timing delay-voice tdm command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). To avoid voice loss at the receiving end of an LMR system, use this command to configure a delay for the voice packet equal to the sum of the durations of all the injected tones and pauses configured with the inject tone command and the inject pause command.
Examples
The following example configures a timing delay of 470 milliseconds before the voice packet is played out:
voice class tone-signal mytonesinject tone 1 1950 3 150inject tone 2 2000 0 60inject pause 3 60inject tone 4 2175 3 150inject tone 5 1000 0 50voice-port 1/0/0voice-class tone-signal mytonestiming delay-voice tdm 470Note that the delay of 470 milliseconds is equal to the sum of the durations of the injected tones and pauses in the tone-signal voice class.
Related Commands
Command Descriptioninject pause
Specifies a pause between injected tones.
inject tone
Specifies a wakeup or frequency selection tone to be played out before the voice packet.
timing hangover
To specify the number of milliseconds of delay before the digital signal processor (DSP) tells Cisco IOS software to turn off the E-lead after the DSP detects that the voice stream has stopped, use the timing hangover command in voice-port configuration mode. To return to the default value, use the no form of this command.
timing hangover milliseconds
no timing hangover milliseconds
Syntax Description
milliseconds
The number of milliseconds for which the E-lead stays active after VAD determines that the voice stream has stopped. Valid values are 0 to 10000. The default is 250 milliseconds.
Defaults
milliseconds: 250 milliseconds
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The timing hangover command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). If the voice port has been configured with the lmr e-lead voice command, use the timing hangover command to adjust the timing if the E-lead is being turned on and off too frequently.
Examples
The following example configures E-lead on voice port 1/0/1 on a Cisco 3745 to stay active for 300 milliseconds after VAD determines that the voice stream has stopped:
voice-port 1/0/1timing hangover 300
timing hookflash-input
To specify the maximum duration of an on-hook condition that will be interpreted as a hookflash by the Cisco IOS software, use the timing hookflash-input command in voice-port configuration mode. To restore the default duration for hookflash timing, use the no form of this command.
timing hookflash-input milliseconds
no timing hookflash-input
Syntax Description
Defaults
milliseconds: 480 milliseconds for E&M voice ports, 1000 milliseconds for FXS voice ports.
Command Modes
Voice-port configuration
Command History
Usage Guidelines
This command is applied to E&M or Foreign Exchange Station (FXS) interfaces.
For Land Mobile Radio E&M voice ports, the timing hookflash-input command configures the delay between when the M-lead is raised and when voice is transmitted. Setting the hookflash duration to 0 milliseconds specifies no delay in the audio input and eliminates front-end clipping.
Analog phones connected to FXS ports use hookflash to access a second dial tone to initiate some phone features, such as transfer and conference. Hookflash is an on-hook condition of short duration that is usually generated when a phone user presses the Flash button on a phone. Cisco voice gateways measure the duration of detected on-hook conditions to determine whether they should be interpreted as hookflash or not. The duration for the on-hook conditions generated by Flash buttons on phones varies for different phone types and is interpreted by Cisco IOS software as follows:
•
An on-hook condition that lasts for a time period that falls inside the hookflash duration range is considered a hookflash.
•
An on-hook condition that lasts for a shorter period than the lower limit of the range is ignored.
•
An on-hook condition that lasts for a longer period than the higher limit of the range is considered a disconnect.
The hookflash duration range for FXS voice ports is defined as follows:
•
The lower limit of the range is set in software at 150 ms, although there is also a hardware-imposed lower limit that is typically about 20 ms, depending on platform type. An on-hook condition that lasts for a shorter time than this hardware-imposed lower limit is simply not reported to the Cisco IOS software.
•
The upper limit of the range is set in software at 1000 ms by default, although this value can be changed using the timing hookflash-input command in voice-port configuration mode on the voice gateway. The upper limit can be set to any value from 50 to 1550 ms. For more information, see the explanations in the "Examples" section.
This command does not affect whether hookflash relay is enabled; hookflash relay is enabled only when the dtmf-relay h245-signal command is configured on the applicable VoIP dial peers. When the dtmf-relay h245-signal command is configured, the H.323 gateway relays hookflash by using an H.245 "signal" User Input Indication method. Hookflash is sent only when an H.245 signal is available.
Examples
The following example sets an upper limit of 200 milliseconds for the hookflash duration range:
voice-port 1/0/0timing hookflash-input 200If the timing hookflash-input command is set to X, a value greater than 150, then any on-hook duration between 150 and X is interpreted as a hookflash. For example, if X is 1550, the hookflash duration range is 150 to 1550 ms. An on-hook signal that lasts for 1250 ms is interpreted as a hookflash, but an on-hook signal of 55 ms is ignored.
voice-port 1/0/0timing hookflash-input 1550If the timing hookflash-input command is set to X, a value less than 150, then any on-hook duration between Y, the hardware lower limit, and X is interpreted as a hookflash. For example, if X is 65, the hookflash duration range is Y to 65 ms. An on-hook signal that lasts for 1250 ms is interpreted as a disconnect, but an on-hook signal of 55 ms is interpreted as a hookflash. (This example assumes that Y for the voice gateway is lower than 55 ms.)
voice-port 1/0/0timing hookflash-input 65Related Commands
Command Descriptiondtmf-relay (Voice over IP)
Specifies how an H.323 gateway relays DTMF tones between telephony interfaces and an IP network.
timing ignore m-lead
To ignore M-lead or voice activity detection (VAD) changes for a specified amount of time after sending the E-lead off signal, use the timing ignore m-lead command in voice-port configuration mode. To return to the default value, use the no form of this command.
timing ignore m-lead milliseconds
no timing ignore m-lead milliseconds
Syntax Description
milliseconds
The number of milliseconds following the sending of the E-lead off signal for which the M-lead and VAD changes are ignored. Valid values are 0 to 10000. The default is 0 milliseconds.
Defaults
milliseconds: 0 milliseconds
Command Modes
Voice-port configuration
Command History
Usage Guidelines
•
The timing ignore m-lead command has an effect on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). Use this command to reduce echo feedback on an LMR voice port. This command has an effect only if the voice port is configured for half duplex mode.
Examples
The following example configures voice port 1/0/1 on a Cisco 3745 to ignore M-lead or VAD changes for 500 milliseconds after sending the E-lead off signal:
voice-port 1/0/1timing ignore m-lead 500
voice class tone-signal
To enter voice-class configuration mode and create a tone-signal voice class, use the voice class tone-signal command in global configuration mode. To delete a tone-signal voice class, use the no form of this command.
voice class tone-signal tag
no voice class tone-signal tag
Syntax Description
Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Usage Guidelines
Use the voice class tone-signal command to define wakeup, frequency selection, and guard tones to be played out before and during the voice packets for a specific voice port. Use the inject guard-tone, inject pause, and inject tone commands to define the tone signaling in this class. You can configure up to ten tones in a tone-signal voice class.
To avoid voice loss at the receiving end of an LMR system, the maximum of the sum of the durations of the injected tones and pauses in the voice class should not exceed 1500 milliseconds. You must also use the timing delay-voice tdm command to configure a delay for the voice packet equal to the sum of the durations of all the injected tones and pauses.
Note that the hyphenation in this command differs from the hyphenation used in a similar command, voice-class tone-signal, which is used in voice-port configuration mode.
Examples
The following example shows how to create a tone-signal voice class starting from global configuration mode:
voice class tone-signal mytonesinject tone 1 1950 3 150inject tone 2 2000 0 60inject pause 3 60inject tone 4 2175 3 150inject tone 5 1000 0 50Related Commands
voice-class tone-signal
To assign a previously configured tone-signal voice class to a voice port, use the voice-class tone-signal command in voice-port configuration mode. To delete a tone-signal voice class, use the no form of this command.
voice-class tone-signal tag
no voice-class tone-signal tag
Syntax Description
tag
Unique label assigned to the voice class. The tag label maps to the tag label created using the voice class tone-signal global configuration command. Can be up to 32 alphanumeric characters.
Defaults
Voice ports have no tone-signal voice class assigned.
Command Modes
Voice-port configuration
Command History
Usage Guidelines
The voice-class tone-signal command is available on an ear and mouth (E&M) voice port only if the signal type for that port is Land Mobile Radio (LMR). Note that the hyphenation in this command differs from the hyphenation used in a similar command, voice class tone-signal, which is used in global configuration mode.
Examples
The following example assigns a previously configured voice class to voice port 1/1/0:
voice-port 1/0/0voice-class tone-signal mytonesRelated Commands
Command Descriptionvoice class tone-signal
Enters voice-class configuration mode and assigns an identification tag number for a tone-signal voice class.
DISCLAIMER
CISCO IOS FEATURES FOR LAND MOBILE RADIO (LMR) OVER IP SHALL BE REFERRED TO HEREINAFTER AS "CISCO LMR FEATURES."
ALL CISCO CUSTOMERS USING THE CISCO LMR FEATURES, ESPECIALLY CUSTOMERS RESPONSIBLE FOR ENSURING PUBLIC SAFETY, ARE STRONGLY ENCOURAGED TO SEEK TECHNICAL SUPPORT FROM A CISCO CERTIFIED SYSTEM INTEGRATOR PARTNER TO ENSURE PROPER CONFIGURATION AND/OR IMPLEMENTATION OF THE CISCO LMR FEATURES INTO THEIR LAND MOBILE RADIO SYSTEMS.
WITH SOLE RESPECT TO THE CISCO LMR FEATURES THEMSELVES, CISCO WILL PROVIDE TECHNICAL SUPPORT IN ACCORDANCE WITH CISCO'S STANDARD POLICIES AND PROCEDURES FOR PROVIDING SUPPORT FOR ANY OTHER CISCO IOS FEATURES. NOTWITHSTANDING THE FOREGOING, IN NO EVENT SHALL CISCO BE HELD RESPONSIBLE FOR PROVIDING ANY TECHNICAL SUPPORT FOR ANY LAND MOBILE RADIO SYSTEMS.
CISCO MAKES NO WARRANTIES, CONDITIONS, OR REPRESENTATION OF ANY KIND, EITHER EXPRESS OR IMPLIED, WITH RESPECT TO ANY LAND MOBILE RADIO SYSTEMS. CISCO SHALL NOT ACCEPT OR ASSUME ANY RESPONSIBILITY OR LIABILITY WITH REGARDS TO ANY THIRD PARTY PRODUCTS OR SERVICES.
Glossary
Applique—Any hardware unit that provides the external interface connections from a router to the network.
COR—Carrier Operated Relay. A signal from a receiver that indicates that the receiver is receiving a signal or carrier and that the receiver is not squelched.
PTT—Push-to-talk. A signal to a radio transmitter that causes the transmission of radio frequency energy.
RTP—Real-Time Transport Protocol. Commonly used with IP networks. RTP is designed to provide end-to-end network transport functions for applications transmitting real-time data, such as audio, video, or simulation data, over multicast or unicast network services. RTP provides such services as payload type identification, sequence numbering, time-stamping, and delivery monitoring to real-time applications.
Squelch—An electric circuit that stops input to a radio receiver when the signal being received is too weak to be anything but noise.
Tone control—The process of sending an in-band tone (2175 Hz) with voice transmission to control receiving radios remotely. In-band tones can be used to control functions such as frequency selection and channel monitoring also. For example, a 1950 Hz tone can be used to select frequency 1, a 1850 Hz tone to select second frequency, or a 2050 Hz tone to activate channel monitor on radio. All of these tones are 40 msec in duration.
VAD—voice activity detection. When enabled on a voice port or a dial peer, silence is not transmitted over the network, only audible speech. When VAD is enabled, the sound quality is slightly degraded but the connection monopolizes much less bandwidth.
VOX—Voice operated transmit. A keying relay that is actuated by sound or voice energy above a certain threshold sensed by a connected acoustoelectric transducer. VOX uses voice energy to key a transmitter, eliminating the need for push-to-talk operation.
Note
Refer to Internetworking Terms and Acronyms for terms not included in this glossary.
Copyright © 2007 Cisco Systems, Inc. All rights reserved.









