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This document provides a consolidated command reference of all the new, changed, and removed commands in Cisco IOS Release 12.0(7)XK.
This document includes the following sections:
The command reference entries in this document are also included in one or more of the following 12.0(7)XK online documents:
This section documents new, modified and removed commands. Modified commands are indicated by an asterisk (*). All other commands used on these platforms are documented in the Cisco IOS Release 12.0 command reference publications.
The following commands have been removed in Cisco IOS Release 12.0(7)XK:
To enable call completion when a PBX does not provide an M-lead response, use the auto-cut-through voice-port configuration command. Use the no form of this command to disable the auto-cut-through operation.
auto-cut-throughThis command has no arguments or keywords.
Auto-cut-through is enabled.
Voice-port configuration
| Release | Modification |
|---|---|
11.3 MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
The auto-cut-through command applies to E&M voice ports only.
The following example enables call completion on a Cisco MC3810 when a PBX does not provide an M-lead response:
router(config)#voice-port 1/1router(config-voiceport)#auto-cut-through
The following example enables call completion on a Cisco 2600 or 3600 when a PBX does not provide an M-lead response:
router(config)#voice-port 1/0/0router(config-voiceport)#auto-cut-through
| Command | Description |
Displays voice port configuration information. |
To specify battery polarity reversal on an FXO or FXS port, use the battery-reversal voice-port configuration command. Use the no form of this command to disable battery reversal.
battery-reversalThis command has no arguments or keywords.
Battery reversal is enabled.
Voice-port configuration
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced. |
The battery-reversal command applies to FXO and FXS voice ports. On Cisco 2600 and 3600 series routers, only analog voice ports in VIC-2FXO-M1 and VIC-2FXO-M2 voice interface cards are able to detect battery reversal; analog voice ports in VIC-2FXO and VIC-2FXO-EU voice interface cards do not detect battery reversal. On digital voice ports, battery reversal is only supported on E1 MELCAS; it is not supported in T1 channel associated signaling (CAS) or E1 CAS.
FXS ports normally reverse battery upon call connection. If an FXS port is connected to an FXO port that does not support battery reversal detection, you can use the no battery-reversal command on the FXS port to prevent unexpected behavior.
FXO ports in loopstart mode normally disconnect calls when they detect a second battery reversal (back to normal). You can use the no battery-reversal command on FXO ports to disable this action.
The battery-reversal command restores voice ports to their default battery-reversal operation.
The following example disables battery reversal on voice port 1/1 on a Cisco MC3810:
router(config)#voice-port 1/1
router(config-voiceport)#no battery-reversal
The following example disables battery reversal on voice port 1/0/0 on a Cisco 2600 or 3600 series router:
router(config)#voice-port 1/0/0
router(config-voiceport)#no battery-reversal
| Command | Description |
Displays voice port configuration information. |
To configure a CCS connection on an interface configured to support CCS frame forwarding, use the ccs connect interface configuration command. To disable the CCS connection on the interface, use the no form of this command.
ccs connect {serial | atm} number [ dlci | pvc vpi/vci | pvc name ] [ cidnumber ]
number | Specify the connection number. |
The following parameters are used for Frame Relay configuration:
serial | Make a serial CCS connection. |
dlci | Specify the DLCI number. |
cidnumber | (Optional) If you have executed the ccs encap frf11 command, the cid option allows you to specify any CID number from 5 to 255. |
atm | Make an ATM CCS connection. |
pvc vpi/vci | Specify the PVC virtual path identifier/virtual channel identifier. Acceptable values are from 0 to 255; the slash is required. |
pvc name | Specify the PVC string that names the PVC for recognition. |
No CCS connection is made.
Serial interface configuration mode
| Release | Modification |
|---|---|
12.0(2)T | This command was introduced for the Cisco MC3810. |
12.0(7)XK | Added CID syntax, removed dlci keyword and vcd options. |
Use this command to configure a CCS connection. If the CCS connection is over Frame Relay, specify a serial interface and the DLCI. If the CCS connection is over ATM, specify atm, the interface number (0 only on the Cisco MC3810), and the PVC.
If you have executed the ccs encap frf11 command, the cidnumber option allows you to specify any CID from 5 to 255. If you do not issue the ccs encap frf11 command, Cisco encapsulation is used, and any CID value other than 254 is ignored.
To configure a frame relay CCS frame-forwarding connection on DLCI 100 by using the default CID of 254, enter the following command:
ccs connect serial 1 100
or:
ccs connect serial 1 100 10
To configure a CCS frame-forwarding connection over an ATM PVC, enter the following command:
ccs connect atm0 pvc 100/10
or:
ccs connect atm0 pvc 10/100 21
or:
ccs connect atm0 pvc mypvc_10 21
To configure a Frame Relay CCS frame-forwarding connection on DLCI 100 using a CID of 110, enter the following command:
ccs connect serial 1 100 110
| Command | Description |
|---|---|
ccs encap frf11 | Allows the specification of the standard Annex-C FRF.11 format. |
To configure the common channel signaling (CCS) packet encapsulation format for FRF.11, use the ccs encap frf11 command. Use the no form of this command to disable ccs encapsulation for FRF11.
ccs encap frf11There are no keywords or arguments.
By default, the format is a Cisco packet format, using a channel ID (CID) of 254.
Serial configuration mode
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced for the Cisco MC3810. |
This command allows the specification of the standard Annex-C format. Use this command to define the packet format for the CCS packet; it places the FRF.11 Annex-C (Data Transfer Syntax) standard header on the CCS packets only.
Once the ccs encap frf11 command is executed, you can use the ccs connect command to specify a CID other than 254.
The following example shows how to configure a serial interface for Frame Relay:
router(config)# interface Serial1:15 router(config-if)# ccs encap frf11 router(config-if)# ccs connect Serial0 990 100
| Command | Description |
|---|---|
mode ccs frame-forwarding | Set to forward frames on the controller. |
To specify the voice codec for a network dial peer, enter the codec dial-peer configuration command. Use the no form of this command to restore the default value.
codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8}[bytes payload-size]
codec | Codec options on the Cisco MC3810 with codec complexity set to high or medium:
|
bytes | (Optional) The voice payload for each frame. |
payload-size | (Optional) Number of bytes you specify as the voice payload of each frame. Acceptable values are from 10 to 240 in increments of 10 (10, 20, 30 ... 220, 230, 240). Any other value is rounded down. |
Dial peers are configured for g729r8.
Dial-peer configuration
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced as a Cisco 3600 VoIP dial-peer configuration command. |
12.0(4)T | This command was modified for VoFR dial peers. On the Cisco MC3810, this command was first supported as a dial-peer command. |
12.0(5)XK and 12.0(7)T | The g729br8 codec and pre-ietf keyword were added for the Cisco 2600 and 3600 platforms. |
12.0(7)XK | The g729abr8 and g729ar8 codecs were added for the Cisco MC3810 and the keyword pre-ietf was deleted. |
A codec type can be configured on the dial-peer if it is supported under the codec complexity setting you have specified.
The dial-peer configuration command is particularly useful when you must change to a small-bandwidth codec. Large-bandwidth codecs, such as G.711, do not fit in a small-bandwidth link. However, g711alaw and g711ulaw provide higher-quality voice transmission than other codecs. For almost toll quality (and a significant savings in bandwidth), g729r8 provides near-toll quality with considerable bandwidth savings.
If the destination router does not support a codec required by the originating router, the call setup fails.
You can change the payload of each voice packet frame by using the bytes payload-size setting. However, increasing the payload size can add processing delay for each voice packet. Table 1 describes the voice payload options and default values for the codecs and packet voice protocols.
| Codec | Protocol | Voice Payload Options (bytes) | Default Voice Payload (bytes) |
|---|---|---|---|
g711alaw | VoIP | 80, 160 | 160 |
g723ar53 | VoIP | 20 to 220 in multiples of 20 | 20 |
g723ar63 | VoIP | 24 to 216 in multiples of 24 | 24 |
g726r16 | VoIP | 20 to 220 in multiples of 20 | 40 |
g726r24 | VoIP | 30 to 210 in multiples of 30 | 60 |
g726r32 | VoIP | 40 to 200 in multiples of 40 | 80 |
g728 | VoIP | 10 to 230 in multiples of 10 | 40 |
g729abr8 | VoIP | 10 to 230 in multiples of 10 | 20 |
The following example configures VoIP dial peer number 10 to use codec type g723r53 (G.723.1 at 5300 bps):
router(config)#dial-peer voice 10 voip
router(config-dialpeer)#codec g723r53
| Command | Description |
codec complexity | This voice-card configuration command sets codec complexity and call density. |
show dial-peer voice | Displays the codec setting for dial peers. |
The codec voice-port configuration command on the Cisco MC3810 is no longer supported beginning in this release. This command was first supported in Cisco IOS Release 11.3(1)MA. Configure the codec value using the codec dial-peer configuration command.
To match the DSP complexity packaging to the codec(s) to be supported, enter the codec complexity voice-card configuration command. The no form of the command restores the default value.
codec complexity {high | medium}
high | With high complexity packaging, each DSP supports two voice channels encoded in any of the following formats: G.711ulaw, G.711alaw, G.723.1(r5.3), G.723.1 Annex A(r5.3), G.723.1(r6.3), G.723.1 Annex A(r6.3), G.726(r16), G.726(r24), G.726(r32), G.729, G.729 Annex B, G.728, and fax relay. |
medium | With medium complexity packaging, each DSP supports four voice channels encoded in any of the following formats: G.711ulaw, G.711alaw, G.726(r16), G.726(r24), G.726(r32), G.729 Annex A, G.729 Annex B with Annex A, and fax relay. This is the default. |
The default is medium complexity.
Voice-card configuration
| Release | Modification |
|---|---|
12.0(5)XK and 12.0(7)T | The command was introduced for the Cisco 2600 and 3600 series. |
12.0(7)XK | This command was first supported on the Cisco MC3810 platform for use with the high performance compression module (HCM). |
Select a higher codec complexity if that is required in order to support a particular codec or combination of codecs.
Select a lower codec complexity to support the greatest number of voice channels, provided that the lower complexity is compatible with the particular codecs in use.
To change codec complexity, all of the DSP voice channels must be in the idle state.
Codec complexity refers to the amount of processing required to perform voice compression. Codec complexity affects the call density—the number of calls that can take place on the digital signal processors (DSPs). With higher codec complexity, fewer calls can be handled.
The following example sets the codec complexity to high on a Cisco MC3810 containing one or two HCMs:
router(config)#voice-card 0
router(config-voicecard)#codec complexity high
The following example sets the codec complexity to high on voice card 1 in a Cisco 2600 or 3600 router:
router(config)#voice-card 1
router(config-voicecard)#codec complexity high
| Command | Description |
show voice dsp | Shows the current status of all DSP voice channels. |
To manipulate the signaling format bit-pattern for all voice signaling types, use the condition command. Use the no form of this command to turn off conditioning on the voice port.
condition {tx-a-bit | tx-b-bit | tx-c-bit | tx-d-bit} {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit} {on | off | invert}
tx-a-bit | Transmit A bit. |
tx-b-bit | Transmit B bit. |
tx-c-bit | Transmit C bit. |
tx-d-bit | Transmit D bit. |
rx-a-bit | Receive A bit. |
rx-b-bit | Receive B bit. |
rx-c-bit | Receive C bit. |
rx-d-bit | Receive D bit. |
on | Forces the bit state to be 1. |
off | Forces the bit state to be 0. |
invert | Inverts the bit state. |
The signaling format is not manipulated (for all transmit or receive A, B, C, and D bits).
Voice-port configuration
| Release | Modification |
|---|---|
11.3 MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
Use the condition command to manipulate the sent or received bit patterns to match expected patterns on a connected device. Be careful not to destroy the information content of the bit pattern. For example, forcing the A-bit on or off will prevent FXO interfaces from being able to generate both an on-hook and off-hook state.
The following example manipulates the signaling format bit-pattern on voice port 1/1 on a Cisco MC3810:
router(config)#voice-port 1/1router(config-voiceport)#condition tx-a-bit invertrouter(config-voiceport)#condition rx-a-bit invert
The following example manipulates the signaling format bit-pattern on voice port 1/1/2 on a Cisco 2600 or 3600:
router(config)#voice-port 1/0/0router(config-voiceport)#condition tx-a-bit invertrouter(config-voiceport)#condition rx-a-bit invert
| Command | Description |
Defines the transmit and receive bits for E&M and E&M MELCAS voice signaling. | |
Configures the E&M or E&M MELCAS voice port to ignore specific receive bits. |
This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
To specify a connection mode for a voice port, use the connection voice-port configuration command. Use the no form of this command to disable the selected connection mode.
connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]}
plar | Specifies a private line automatic ring down (PLAR) connection. PLAR is an autodialing mechanism that permanently associates a voice interface with a far-end voice interface, allowing call completion to a specific telephone number or PBX without dialing. When the calling telephone goes off hook a predefined network dial peer is automatically matched, which sets up a call to the destination telephone or PBX. |
tie-line | Specifies a connection that emulates a temporary tie-line trunk to a private branch exchange (PBX). A tie-line connection is automatically set up for each call and torn down when the call ends. |
plar-opx | Specifies a PLAR Off-Premises eXtension connection. Using this option, the local voice port provides a local response before the remote voice port receives an answer. On FXO interfaces, the voice port will not answer until the remote side answers. |
trunk | Specifies a connection that emulates a permanent trunk connection to a private branch exchange (PBX). A trunk connection remains "nailed up" in the absence of any active calls. |
digits | Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number. |
answer-mode | (Optional; used only with the trunk keyword.) Specifies that the router should not attempt to initiate a trunk connection, but should wait for an incoming call before establishing the trunk. |
No connection mode is specified.
Voice-port configuration
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced. |
11.3(1)MA1 | This command was first supported on the Cisco MC3810, and the tie-line keyword was first made available on the Cisco MC3810. |
11.3(1)MA5 and 12.0(2)T | The plar-opx keyword was first made available on the Cisco MC3810 as the plar-opx-ringrelay keyword. The keyword was shortened in a subsequent release. |
12.0(3)XG | The trunk keyword was made available on the Cisco MC3810. The trunk answer-mode option was added. |
12.0(7)XK | This command was unified across the Cisco 2600, 3600, and MC3810 platforms. |
Use this command to specify a connection mode for a specific interface. For example, use the connection plar command to specify a PLAR interface. The string you configure for this command is used as the called number for all incoming calls over this connection. The destination peer is determined by the called number.
Use the connection trunk command to specify a permanent, "nailed up" tie-line connection to a PBX. You can use the connection trunk command for E&M-to-E&M trunks, FXO-to-FXS trunks, and FXS-to-FXS trunks. Signaling will be transported for E&M-to-E&M trunks and FXO-to-FXS trunks; signaling will not be transported for FXS-to-FXS trunks.
To configure one of the devices in the trunk connection to act as slave and only receive calls, use the answer-mode option with the connection trunk command when configuring that device.
Use the connection tie-line command when the dial plan requires that additional digits be added in front of any digits dialed by the PBX, and that the combined set of digits be used to route the call onto the network. The operation is similar to the connection plar command operation, but in this case the tie-line port waits to collect digits from the PBX. The tie-line digits are automatically stripped by a terminating port.
If the connection command is not configured, the standard session application outputs a dial tone when the interface goes off-hook until enough digits are collected to match a dial-peer and complete the call.
The following example selects PLAR as the connection mode on a Cisco 3600, with a destination telephone number of 555-9262:
router(config)#voice-port 1/0/0router(config-voiceport)#connection trunk 5559262
The following example selects tie-line as the connection mode on a Cisco MC3810, with a destination telephone number of 555-9262:
router(config)#voice-port 1/1router(config-voiceport)#connection tie-line 5559262
The following example specifies a PLAR off-premises extension connection on a Cisco 3600, with a destination telephone number of 555-9262:
router(config)#voice-port 1/0/0router(config-voiceport)#connection plar-opx 5559262
The following example configures a Cisco 3600 series router for a trunk connection and specifies that it will establish the trunk only when it receives an incoming call:
router(config)#voice-port 1/0/0router(config-voiceport)#connection trunk 5559262 answer-mode
| Command | Description |
session-protocol | Establishes a session protocol for calls between the local and remote routers via the packet network. |
session-target | Configures a network-specific address for a dial peer. |
dial-peer voice | Enters dial-peer configuration mode and specifies the method of voice-related encapsulation. |
destination-pattern | Specifies either the prefix or the full E.164 telephone number to be used for a dial peer. |
To define the transmit and receive bits for E&M and E&M Mercury Exchange Limited (MELCAS) voice signaling, use the define voice-port configuration command. Use the no form of this command to restore the default value.
define {Tx-bits | Rx-bits} {seize | idle} {0000 | 0001 | 0010 | 0011 | 0100 | 0101 |
Tx-bits | Transmit signaling bits. |
Rx-bits | Receive signaling bits. |
seize | The bit pattern defines the seized state. |
idle | The bit pattern defines the idle state. |
0000 through 1111 | Specifies the bit pattern. |
The default is to use the preset signaling patterns as defined in ANSI and CEPT standards, as follows:
For 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
Voice-port configuration
| Release | Modification |
|---|---|
11.3(1) MA3 | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
This 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.
To configure a voice port on a Cisco 2600 or 3600 router sending traffic in North American E&M signaling format to convert the signaling to MELCAS format, enter the following commands:
router(config)#voice-port 1/0/0router(config-voiceport)#define rx-bits idle 1101router(config-voiceport)#define rx-bits idle 0101router(config-voiceport)#define tx-bits seize 1101router(config-voiceport)#define tx-bits seize 0101
To configure a voice port on a Cisco MC3810 sending traffic in North American E&M signaling format to convert the signaling to MELCAS format, enter the following commands:
router(config)#voice-port 0/8router(config-voiceport)#define rx-bits idle 1101router(config-voiceport)#define rx-bits idle 0101router(config-voiceport)#define tx-bits seize 1101router(config-voiceport)#define tx-bits seize 0101
| Command | Description |
Manipulate the signaling bit-pattern for all voice signaling types. | |
Configures an E&M or E&M MELCAS voice port to ignore specific receive bits. |
To specify a hunt selection order for dial-peers, use the dial-peer hunt dial-peer configuration command. Use the no form of this command to restore the default selection order.
dial-peer hunt hunt-order-number
hunt-order-number | A number from 0 to 7 that selects a predefined hunting selection order: 0—Longest match in phone number, explicit preference, random selection. This is the default hunt order number. 1—Longest match in phone number, explicit preference, least recent use. 2—Explicit preference, longest match in phone number, random selection. 3—Explicit preference, longest match in phone number, least recent use. 4—Least recent use, longest match in phone number, explicit preference. 5—Least recent use, explicit preference, longest match in phone number. 6—Random selection. 7—Least recent use. |
The default is the longest match in phone number, explicit preference, and random selection (hunt order number 0).
Global configuration
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced, and was first supported on the Cisco 2600 and 3600 series routers and on the Cisco MC3810 multiservice access concentrator. |
Use the dial-peer hunt dial-peer configuration command if you have configured hunt groups. "Longest match in phone number" refers to the destination pattern that matches the greatest number of the dialed digits. "Explicit preference" refers to the preference setting in the dial-peer configuration. "Least recent use" refers to the destination pattern that has waited the longest since being selected. "Random selection" weighs all of the destination patterns equally in a random selection mode.
The following example configures the dial peers to hunt in the following order: (1) longest match in phone number, (2) explicit preference, (3) random selection.
router# configure terminal router(config)# dial-peer hunt 0
| Command | Description |
preference | Specifies the preferred selection order of a dial peer within a hunt group. |
destination-pattern | Specifies the prefix or the complete telephone number for a dial peer. |
show dial-peer voice | Displays configuration information for dial peers. |
To change the character used as a terminator for variable length dialed numbers, use the dial-peer terminator global configuration command. Use the no form of this command to restore the default terminating character.
dial-peer terminator character
character | Designates the terminating character for a variable-length dialed number. Valid numbers and characters are #, *, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, a, b, c, and d. The default is #. |
The default terminating character is #.
Global configuration
| Release | Modification |
|---|---|
12.0 | This command was introduced. |
12.0(7)XK | Usage was restricted to variable-length dialed numbers. |
There are certain areas in the world, 0for example, in certain European countries, where telephone numbers can vary in length. When a dialed-number string is identified as a variable length dialed-number, the system does not place a call until the configured value for the timeouts interdigits command has expired, or until the caller dials the terminating character. Use the dial-peer terminator global configuration command to change the terminating character.
The following example specifies "9" as the terminating character for variable-length dialed numbers:
router# configure terminal router(config)# dial-peer terminator 9#
| Command | Description |
answer-address | Specifies the preferred selection order of a dial peer within a hunt group. |
destination-pattern | Specifies the prefix or the complete telephone number for a dial peer. |
timeouts interdigit | Specifies the interdigit timeout value for a voice port in seconds. |
show dial-peer voice | Displays configuration information for dial peers. |
To enter dial-peer configuration mode and specify the method of voice encapsulation, use the dial-peer voice global configuration command.
For the Cisco 2600 series:
dial-peer voice tag {pots | voip | vofr}For the Cisco 3600 series and the Cisco MC3810:
dial-peer voice tag {pots | voip | vofr | voatm}
tag | A number identifying a particular dial peer. Valid entries are 1 to 2147483647. |
pots | POTS dial peer using basic telephone service. |
voip | VoIP dial peer using voice encapsulation on the POTS network. |
vofr | Voice over Frame Relay dial peer using encapsulation on the Frame Relay backbone network. |
voatm | (Cisco 3600 and MC3810 only) Voice over ATM dial peer using real-time AAL5 voice encapsulation on the ATM backbone network. |
No default behavior or values.
Global configuration
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced. |
11.3(1)MA | This command was first supported on the Cisco MC3810 with support for POTS, VoFR and VoATM. |
12.0(3)XG | This command added VoFR to the Cisco 2600 and 3600 series routers. |
12.0(4)T | This command added VoFR to the Cisco 7200 series platform. |
12.0(7)XK | This command added VoIP to the Cisco MC3810 and VoATM to the Cisco 3600 series routers. Support for VoHDLC on the Cisco MC3810 was removed in this release. |
Use the dial-peer voice global configuration command to switch to the dial-peer configuration mode from the global configuration mode. Use the exit command to exit the dial-peer configuration mode and return to the global configuration mode.
The following example accesses dial-peer configuration mode and configures a POTS peer identified as dial peer 10:
router# configure terminal router(config)# dial-peer voice 10 pots
The following example accesses dial-peer configuration mode and configures a VoATM peer identified as dial peer 20:
router# configure terminal router(config)# dial-peer voice 20 voatm
| Command | Description |
voice-port | Enters voice-port configuration mode. |
To configure an FXS voice port to return an acknowledgment upon receipt of a disconnect signal, use the disconnect-ack voice-port configuration command. To disable the acknowledgment, use the no form of this command.
disconnect-ackThis command has no arguments or keywords.
FXS voice ports return an acknowledgment upon receipt of a disconnect signal.
Voice-port configuration
| Release | Modification |
|---|---|
11.3 MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
This command configures an FXS voice port to remove line power if the equipment on an FXS loop-start trunk disconnects first.
The following example turns off the disconnect acknowledgment signal on voice port 1/1 on a Cisco MC3810:
router(config)#voice-port 1/1router(config-voiceport)#no disconnect-ack
The following example turns off the disconnect acknowledgment signal on voice port 1/1/0 on a Cisco 2600 or 3600:
router(config)#voice-port 1/0/0router(config-voiceport)#no disconnect-ack
| Command | Description |
Displays voice port configuration information. |
To specify the DS0 timeslots that make up a logical voice port on a T1 or E1 controller, and to specify the signaling type, use the ds0-group controller configuration command. Use the no form of the command to remove the DS0 group and signaling setting.
ds0-group ds0-group-no timeslots timeslot-list type signal-type
no ds0-group ds0-group-no
ds0-group-no | A value from 0 to 23 that identifies the DS0 group. |
timeslot-list | timeslot-list is a single timeslot number, a single range of numbers, or multiple ranges of numbers separated by commas. For T1, allowable values are from 1 to 24. Examples are: · 2 · 1-15, 17-24 · 1-23 · 2, 4, 6-12 |
type | The signaling method selection for type depends on the connection that you are making. The E&M interface allows connection for PBX trunk lines (tie- lines) and telephone equipment. The FXS interface allows connection of basic telephone equipment and PBXs. The FXO interface is for connecting the central office (CO) to a standard PBX interface where permitted by local regulations. The FXO interface is often used for off-premises extensions. The options are as follows: · e&m-immediate-start—no specific off-hook and on-hook signaling · e&m-delay-dial—the originating endpoint sends an off-hook signal and then waits for an off-hook signal followed by an on-hook signal from the destination · e&m-wink-start—the originating endpoint sends an off-hook signal and waits for a wink signal from the destination · fxs-ground-start—Foreign Exchange Station ground-start signaling support · fxs-loop-start —Foreign Exchange Station loop-start signaling support · fxo-ground-start—Foreign Exchange Office ground-start signaling support · fxo-loop-start—Foreign Exchange Office loop-start signaling support |
| The following options are available only on E1 controllers on the Cisco MC3810: · e&m-melcas-immed—E&M Mercury Exchange Limited Channel Associated Signaling (MELCAS) immediate start signaling support · e&m-melcas-wink—E&M MELCAS wink start signaling support · e&m-melcas-delay—E&M MELCAS delay start signaling support · fxo-melcas—MELCAS Foreign Exchange Office signaling support · fxs-melcas—MELCAS Foreign Exchange Station signaling support |
| The ext-sig option is available only when the mode ccs command is enabled on the Cisco MC3810 for FRF.11 transparent CCS support. |
No DS0 group is defined.
Controller configuration
| Release | Modification |
|---|---|
11.2 | This command was introduced for the Cisco AS5300 as cas-group. |
12.0(1)T | The cas-group command was first supported on the Cisco 3600 series. |
12.0(5)T | This command was renamed ds0-group on the Cisco AS5300 and on the Cisco 2600 and 3600 series (requires Digital T1 Packet Voice Trunk Network Modules). |
12.0(7)XK | Support for this command was extended to the Cisco MC3810. When the ds0-group command became available on the Cisco MC3810, the voice-group command was removed and no longer supported. ext-sig replaces the ext-sig-master and ext-sig-master options that were available with the voice-group command. |
The ds0-group command automatically creates a logical voice port that is numbered as follows:
Cisco 2600 and 3600 series:
slot/port:ds0-group-no.
Cisco MC3810:
slot:ds0-group-no
On the Cisco MC3810, the slot number is the controller number. Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
The following example configures ranges of T1 controller timeslots for FXS ground-start and FXO loop-start signaling on a Cisco 2600 or 3600 Series router:
router(config)# controller T1 1/0 router(config-controller)# framing esf router(config-controller)# linecode b8zs router(config-controller)# ds0-group 1 timeslot 1-10 type fxs-ground-start router(config-controller)# ds0-group 2 timeslot 11-24 type fxo-loop-start
The following example configures DS0 groups 1 and 2 on controller T1 1 on the Cisco MC3810 to support Transparent CCS:
router(config)#controller T1 1router(config-controller)#mode ccs cross-connectrouter(config-controller)#ds0-group 1 timeslot 1-10 type ext-sigrouter(config-controller)#ds0-group 2 timeslot 11-24 type ext-sig
| Command | Description |
codec complexity | Matches the DSP complexity packaging to the codec(s) to be supported |
mode ccs | Configures the T1/E1 controller to support CCS cross-connect or CCS frame-forwarding. |
aal-encap | ATM adaptation layer (AAL) and encapsulation type. Possible values for aal-encap are as follows: · aal5mux voice—For a MUX-type virtual circuit for Voice over ATM. · aal5snap—The only encapsulation supported for Inverse ARP. Logical Link Control/Subnetwork Access Protocol (LLC/SNAP) precedes the protocol datagram. |
The global default encapsulation is aal5snap. See the "Usage Guidelines" section for other default characteristics.
Interface-ATM-VC configuration (for an ATM PVC or SVC)
| Release | Modification |
|---|---|
11.3 T | This command was introduced. |
12.0 | This command superseded the encapsulation atm command for the Cisco MC3810, and the aal5mux frame and aal5mux voice suboptions appeared. |
12.0(7)XK | Support for the aal5mux voice option was added to the Cisco 3600 series routers. |
Use one of the aal5mux encapsulation options to dedicate the specified PVC to a single protocol; use the aal5snap encapsulation option to multiplex two or more protocols over the same PVC. Whether you select aal5mux or aal5snap encapsulation depends on practical considerations, such as the type of network and the pricing offered by the network. If the network's pricing depends on the number of PVCs set up, aal5snap may be the appropriate choice. If pricing depends on the number of bytes transmitted, aal5mux may be the appropriate choice because it has slightly less overhead.
If you specify virtual template parameters after the ATM PVC is configured, issue a shutdown command followed by a no shutdown command on the ATM subinterface to restart the interface, causing the newly configured parameters (such as an IP address) to take effect.
The following example configures a PVC to support encapsulation for Voice over ATM:
router(config-if)# pvc 20 router(config-if-atm-pvc)# encapsulation aal5mux voice
This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
To specify which digits to forward for voice calls, use the forward-digits dial-peer configuration command. If the no form of this command is entered, any digits not matching the destination-pattern are not forwarded. Use the default form of this command to restore the default state.
forward-digits {num-digit | all | extra}
num-digit | The number of digits to be forwarded. If the number of digits is greater than the length of a destination phone number, the length of the destination number is used. The valid range is 0 to 32. Setting the value to 0 is equivalent to entering no forward-digits. |
all | Forward all digits. If all is entered, the full length of the destination pattern is used. |
extra | If the length of the dialed digit string is greater than the length of the dial-peer destination pattern, the extra right-justified digits are forwarded. However, if the dial-peer destination pattern is variable length (ending with character "T", for example: T, 123T, 123...T), extra digits are not forwarded. |
Dialed digits not matching the destination-pattern are forwarded.
Dial-peer configuration
| Release | Modification |
|---|---|
11.3(1) MA | This command was introduced on the Cisco MC3810. |
12.0(2) T | The implicit option was added. |
12.0(4) T | This command was modified to support ISDN PRI QSIG signaling calls. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 platforms, the implicit keyword was removed, and the extra keyword was added. |
This command applies only to POTS dial peers.
Forwarded digits are always right-justified, so that extra leading digits are stripped.
The destination pattern includes both explicit digits and wildcards if present.
Use the default form of this command if a non-default digit-forwarding scheme was entered previously and you wish to restore the default.
For QSIG ISDN connections, entering forward-digits all implies that all the digits of the called party number are sent to the ISDN connection. When you enter forward-digits num-digit and enter a number from 1 to 32, the number of digits specified (right justified) of the called part number are sent to the ISDN connection.
The following example forwards all of the digits in the destination pattern of a POTS dial peer:
router(config)# dial-peer voice 1 pots
router(config-dial-peer)# destination-pattern 8...
router(config-dial-peer)# forward-digits all
The following example forwards 4 of the digits in the destination pattern of a POTS dial peer:
router(config)# dial-peer voice 1 pots
router(config-dial-peer)# destination-pattern 555....
router(config-dial-peer)# forward-digits 4
The following example forwards the extra right-justified digits that exceed the length of the destination pattern of a POTS dial peer:
router(config)# dial-peer voice 1 pots
router(config-dial-peer)# destination-pattern 555....
router(config-dial-peer)# forward-digits extra
| Command | Description |
destination-pattern | Defines the prefix or the full E.164 telephone number to be used for a dial peer. |
show dial-peer voice | Displays configuration information for dial peers. |
This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
To assign a data link connection identifier (DLCI) to a specified Frame Relay subinterface on the router or access server, use the frame-relay interface-dlci interface configuration command. Use the no form of this command to remove this assignment.
frame-relay interface-dlci dlci [ietf | cisco] [voice-cir cir]
dlci | DLCI number to be used on the specified subinterface. |
ietf | cisco | (Optional) Encapsulation type: Internet Engineering Task Force (IETF) Frame Relay encapsulation or Cisco Frame Relay encapsulation. |
voice-cir cir | (Optional; supported on the Cisco MC3810 only.) Specifies the upper limit on the voice bandwidth that may be reserved for this DLCI. The default is the CIR configured for the Frame Relay map class. For more information, see the "Usage Guidelines" section. |
No DLCI is assigned.
Interface configuration
| Release | Modification |
|---|---|
10.0 | This command was introduced. |
11.3(1) MA | The voice-encap option was added for the Cisco MC3810. |
12.0(2) T | The voice-cir option was added for the Cisco MC3810. |
12.0(3)XG and 12.0(4)T | Additional usage guidelines added. |
12.0(7)XK | The voice-encap option on the Cisco MC3810 was removed. This option is no longer supported. |
This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
This command was added in Cisco IOS Release 12.0(2)T on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
To disable all further dial-peer hunting if a call fails when using hunt groups, enter the huntstop dial-peer configuration command. To reenable dial-peer call hunting, enter the no form of this command.
huntstopThis command has no arguments or keywords.
Disabled
Dial-peer configuration
| Release | Modification |
|---|---|
12.0(5)T | This command was introduced on the Cisco MC3810. |
12.0(7)XK | Support for this command was extended to the Cisco 2600 and 3600 series routers. |
Once you enter this command, no further hunting is allowed if a call fails on the specified dial peer.
This command can be used with all types of dial peers.
The following example shows how to disable dial-peer hunting on a specific dial peer:
router(config)# dial peer voice 100 vofr
router(config-dial-peer)# huntstop
The following example shows how to reenable dial-peer hunting on a specific dial peer:
router(config)# dial peer voice 100 vofr
router(config-dial-peer)# no huntstop
| Command | Description |
dial-peer voice | Enters dial-peer configuration mode and specifies the method of voice-related encapsulation. |
number | Integer, expressed in equipment impairment factor units, specifying the ICPIF value. Valid entries are from 0 to 55. |
The default value for this command is 30.
Dial-peer configuration
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced on the Cisco 3600 series. |
12.0(7)XK | This command was first supported on the Cisco MC3810 platform. |
Use the icpif command to specify the maximum acceptable impairment factor for the voice calls sent by the selected dial peer.
This command is applicable only to VoIP peers.
The following example disables the icpif command:
dial-peer voice 10 voip icpif 0
To configure the E&M or E&M MELCAS voice port to ignore specific receive bits, use the ignore voice-port configuration command. Use the no form of this command to restore the default value.
ignore {rx-a-bit | rx-b-bit | rx-c-bit | rx-d-bit}
rx-a-bit | Ignores the receive A bit. |
rx-b-bit | Ignores the receive B bit. |
rx-c-bit | Ignores the receive C bit. |
rx-d-bit | Ignores the receive D bit. |
The default is mode-dependent:
E&M:
no ignore rx-a-bit
ignore rx-b-bit, rx-c-bit, rx-d-bit
E&M MELCAS:
no ignore rx-b-bit, rx-c-bit, rx-d-bit
Voice-port configuration
| Release | Modification |
|---|---|
11.3 MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
This command applies to E&M digital voice ports associated with T1/E1 controllers. Repeat the command for each receive bit to be configured. Use this command with the define command.
To configure voice-port 1/1 on a Cisco MC3810 to ignore receive bits a, b, and c and to monitor receive bit d, enter the following commands:
router(config)#voice-port 1/1router(config-voiceport)#ignore rx-a-bitrouter(config-voiceport)#ignore rx-b-bitrouter(config-voiceport)#ignore rx-c-bitrouter(config-voiceport)#no ignore rx-d-bit
To configure voice-port 1/0/0 on a Cisco 3600 to ignore receive bits a, c, and d and to monitor receive bit b, enter the following commands:
router(config)#voice-port 1/0/0router(config-voiceport)#ignore rx-a-bitrouter(config-voiceport)#ignore rx-c-bitrouter(config-voiceport)#ignore rx-d-bitrouter(config-voiceport)#no ignore rx-b-bit
| Command | Description |
Manipulates the signaling bit-pattern for all voice signaling types. | |
Defines the transmit and receive bits for E&M and E&M MELCAS voice signaling. | |
show voice port | Displays configuration information for voice ports. |
string | Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number. |
The default value for this command is no associated called number.
Dial peer configuration
| Release | Modification |
|---|---|
11.3NA | This command was introduced on the Cisco AS5800 platform. |
12.0(7)XK | This command was first supported on the Cisco MC3810 platform. |
When the Cisco MC3810 is handling both modem and voice calls, it needs to be able to identify the service type of the call—meaning whether the incoming call to the server is a modem or a voice call. When the access server handles only modem calls, the service type identification is handled through modem pools. Modem pools associate calls with modem resources based on the called number (DNIS). In a mixed environment, where the server receives both modem and voice calls, you need to identify the service type of a call by using the incoming called-number command.
If you do not use the incoming called-number command, the server attempts to resolve whether an incoming call is a modem or voice call based on the interface over which the call comes. If the call comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes in over a voice port associated with a dial peer, the call is assumed to be a voice call.
By default, there is no called number associated with the dial peer, which means that incoming calls will be associated with dial peers based on matching calling number with answer address, call number with destination pattern, or calling interface with configured interface.
This command applies to both VoIP and POTS dial peers.
The following example configures calls coming in to the server with a called number of "3799262" as voice calls:
dial peer voice 10 pots incoming called-number 3799262
To configure contiguous bearer channel handling on an E1 Primary Rate Interface (PRI) interface, use the isdn contiguous-bchan interface configuration command. To disable the contiguous B channel handling, use the no form of this command.
isdn contiguous-bchanThis command has no arguments or keywords.
By default, contiguous B channel handling is off.
Interface configuration
| Release | Modification |
12.0(7)XK | This command was introduced. |
Use the isdn contiguous-bchan command to specify contiguous bearer channel handling so that B channels 1 through 30, skipping 16, map to timeslots 1 through 31). This is available for E1 PRI interfaces only, when the primary-qsig switch type option is configured by using the isdn switch-type command.
The following example shows the command configuration on a Cisco 3660 series router E1 interface:
interface Serial5/0:15 no ip address ip mroute-cache no logging event link-status isdn switch-type primary-qsig isdn overlap-receiving isdn incoming-voice voice isdn continguous-bchan
| Command | Description |
|---|---|
isdn switch-type primary-qsig | In global or interface configuration mode, configures the primary-qsig switch type for PRI support. |
To route all incoming voice calls as voice calls, to route them the modem and treat them as analog data, or to ensure that calls bypass the modems and are treated as digital data, use the isdn incoming-voice interface configuration command. Use the no form of this command to disable the setting.
isdn incoming-voice {data [56 | 64] | modem [56 | 64] | voice}
data | Specifies that incoming voice calls bypass the modems and are handled as digital data. |
modem | Specifies that incoming voice calls are passed over to the digital modems, where they negotiate the appropriate modem connection with the far-end modem. |
voice | Specifies that incoming voice calls are treated as voice calls rather than being routed to the modem or handled as digital data. |
56 | Specifies that the bandwidth for this connection is 56 kbps. |
64 | Specifies that the bandwidth for this connection is 64 kbps. If no argument is entered for either the data or modem keywords, the default value is 64. |
When a PRI or BRI interface is created, isdn incoming-voice voice is the default, except on a Cisco 2600 or 3600 BRI S/T TE interface. In this case, if the command is not specified, the default isdn incoming-voice modem configuration setting is converted to isdn incoming-voice voice when the interface receives an incoming call.
Interface configuration
| Release | Modification |
11.1 | This command was introduced. |
12.0(2)XC and 12.0(3)T | This command was made available for BRI interfaces. |
12.0(7)XK | This command was modified to include the voice keyword. |
Unless you specify otherwise, all calls received by the router and characterized as voice calls are treated as such and not handled as digital data or not passed over to the modem.
On a Cisco 2600 or 3600 series router BRI S/T TE interface where the isdn incoming-voice command is not specified, the default isdn incoming-voice modem configuration setting is converted to isdn incoming-voice voice when the interface receives an incoming call.
To establish speedier connections for analog calls to the router, use the isdn incoming-voice command with the modem keyword to have voice calls routed through digital modems (as pulse-code modulated analog data) instead of being treated as digital data.
The following example shows the command configuration on a Cisco 3660 series router T1 PRI interface:
interface Serial5/0:23 no ip address ip mroute-cache no logging event link-status isdn switch-type primary-qsig isdn overlap-receiving isdn incoming-voice voice
To configure a Primary Rate Interface (PRI) interface to serve as either the primary QSIG slave or the primary QSIG master, use the isdn protocol-emulate interface command. To disable QSIG signaling, use the no form of this command.
isdn protocol-emulate { user | network }
user | Enter user (equivalent to the QSIG term slave) to configure the port as the terminating end. This is the default. |
network | Enter network (equivalent to the QSIG term master) to configure the port as NT; the PINX is the slave. |
User
Interface configuration mode.
| Release | Modification |
12.0(7)T | This command was introduced for the Cisco AS5300. |
12.0(7)XK | This command was introduced for the Cisco MC3810, and for the Cisco 7200 VXR, Cisco 2600, and Cisco 3600 series routers. |
On the Cisco MC3810, this command replaces the command isdn switch-type [primary-qsig-slave | primary-qsig-master] command.
The following example shows the command configuration on a Cisco 3660 series router T1 PRI interface:
interface Serial5/0:23 no ip address ip mroute-cache no logging event link-status isdn switch-type primary-qsig isdn overlap-receiving isdn protocol-emulate user
To specify a central office switch type or configure a Primary Rate Interface (PRI) interface to support Q.SIG signaling, use the isdn switch-type global or interface command. To disable the central office switch type or QSIG signaling, use the no form of this command.
isdn switch-type {switch-type | primary-qsig | basic-qsig}
switch-type | Service provider switch type. See Table 2 for a list. |
primary-qsig | PRI |
basic-qsig | BRI |
The switch type defaults to none, which disables the switch type.
Global configuration mode or interface configuration mode.
| Release | Modification |
|---|---|
9.21 | Introduced as a global command. |
11.3 T | Introduced as an interface command. |
12.0(2)T | primary-qsig-slave and primary-qsig-master keywords introduced for the Cisco MC3810. |
12.0(7)K | primary-qsig-slave and primary-qsig-master keywords for the Cisco MC3810 are no longer supported. primary-qsig and basic-qsig keywords supported on the Cisco MC3810, Cisco 7200 VXR, 2600 and 3600 series routers. |
You can enter the isdn switch-type command to support QSIG at either the global configuration level or at the interface configuration level. For example, if you have a QSIG connection on one line as well as on the BRI or PRI port, you can configure the ISDN switch type in one of the following combinations:
| Country | ISDN Switch Type | Description |
|---|---|---|
Australia | basic-ts013 | Australian TS013 switches |
Europe | basic-1tr6 | German 1TR6 ISDN switches |
| basic-nwnet3 | Norwegian NET3 ISDN switches (phase 1) |
| basic-net3 | NET3 ISDN switches (UK and others) |
| vn2 | French VN2 ISDN switches |
| vn3 | French VN3 ISDN switches |
Japan | ntt | Japanese NTT ISDN switches |
New Zealand | basic-nznet3 | New Zealand NET3 switches |
North America | basic-5ess | Lucent Technologies basic rate switches |
| basic-dms100 | NT DMS-100 basic rate switches |
| basic-ni1 | National ISDN-1 switches |
The following example shows the command configuration on a Cisco 3660 series router T1 PRI interface:
interface Serial5/0:23 no ip address ip mroute-cache no logging event link-status isdn switch-type primary-qsig isdn overlap-receiving isdn protocol-emulate user
| Command | Description |
|---|---|
Configures the interface to serve as either the QSIG slave or the QSIG master. |
To specify the analog-to-digital gain offset for an analog FXO or FXS voice port, enter the codec dial-peer configuration command. Use the no form of this command to restore the default value.
loss-plan {plan1 | plan2 | plan3 | plan4 | plan5 | plan6 | plan7 | plan8 | plan9}
plan1 | FXO: A-D gain = 0 dB, D-A gain = 0 dB FXS: A-D gain = -3 dB, D-A gain = -3 dB |
plan2 | FXO: A-D gain = 3 dB, D-A gain = 0 dB FXS: A-D gain = 0 dB, D-A gain = -3 dB |
plan3 | FXO: A-D gain = -3 dB, D-A gain = 0 dB FXS: Not applicable |
plan4 | FXO: A-D gain = -3 dB, D-A gain = -3 dB FXS: Not applicable |
plan5 | FXO: Not applicable FXS: A-D gain = -3 dB, D-A gain = -10 dB |
plan6 | FXO: Not applicable FXS: A-D gain = 0 dB, D-A gain = -7 dB |
plan7 | FXO: A-D gain = 7 dB, D-A gain = 0 dB FXS: A-D gain = 0 dB, D-A gain = -6 dB |
plan8 | FXO: A-D gain = 5 dB, D-A gain = -2 dB FXS: Not applicable |
plan9 | FXO: A-D gain = 6 dB, D-A gain = 0 dB FXS: Not applicable |
FXO: A-D gain = 0 dB, D-A gain = 0 dB (loss plan 1)
FXS: A-D gain = -3 dB, D-A gain = -3 dB (loss plan 1)
Voice-port configuration
| Release | Modification |
|---|---|
11.3(1)MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | The following additional signal level choices were added: plan 3, plan 4, plan 8, and plan 9. |
This command sets the analog signal level difference (offset) between the analog voice port and the digital signal processor (DSP). Each loss plan specifies a level offset in both directions—from the analog voice port to the DSP (A-D) and from the DSP to the analog voice port (D-A).
Use this command to obtain the required levels of analog voice signals to and from the DSP.
This command is supported only on Cisco MC3810 series concentrators, on FXO and FXS analog voice ports.
The following example configures FXO voice port 1/6 for a -3 dB offset from the voice port to the DSP and a 0 dB offset from the DSP to the voice port:
router(config)#voice-port 1/6
router(config-voiceport)#loss-plan plan3
The following example configures FXS voice port 1/1 for a 0 dB offset from the voice port to the DSP and a -7 dB offset from the DSP to the voice port:
router(config)#voice-port 1/1
router(config-voiceport)#loss-plan plan6
| Command | Description |
impedance | Specifies the terminating impedance of the voice port interface. Used on FXO voice ports in correcting input levels. |
input gain | Specifies the gain applied by a voice port to the input signal from the PBX or other customer premises equipment. |
output attenuation | Specifies the attenuation applied by a voice port to the output signal toward the PBX or other customer premises equipment. |
To define a complete telephone number for an extension, use the num-exp global configuration command. Use the no form of this command to cancel a configured number expansion.
num-exp extension-number expanded-number
extension-number expanded-number | Digit(s) defining an extension number to be expanded. Digit(s) defining the expanded telephone number or destination pattern. |
No number expansion is configured.
Global configuration
| Release | Modification |
|---|---|
11.3(1)T | This command was first introduced on the Cisco 3600 platform. |
12.0(3)T | This command was first supported on the Cisco AS5300 platform. |
12.0(4)XL | This command was first supported on the Cisco AS5800 platform. |
12.0(7)XK | This command was first supported on the Cisco MC3810 platform. |
Use the num-exp global configuration command to expand a set of numbers (for example, an extension number) into a destination pattern. With this command, you can map specific extensions and expanded numbers together by explicitly defining each number, or you can define extensions and expanded numbers using variables. You can also use this command to convert seven-digit numbers to numbers containing less than seven digits.
Use a period (.) as a variable or wild card, representing a single number. Use a separate period for each number you want to represent with a wildcard; if you want to replace four numbers in an extension with wildcards, type in four periods.
The following example specifies that extension number 55541 be expanded to 14085555541:
num-exp 55541 14085555541
The following example specifies that all five-digit extensions beginning with 5 be expanded to 1408555 . . . .
num-exp 5.... 1408555....
| Command | Description |
forward-digits | Specifies which digits to forward for voice calls. |
prefix | Specifies a prefix for a dial peer. |
dial-peer terminator | Change the character used as a terminator for variable length dialed numbers. |
To tune the playout buffer to accommodate packet jitter caused by switches in the WAN, use the playout-delay voice-port configuration command. Use the no form of this command to restore the default value.
playout-delay {maximum | nominal} milliseconds
maximum | The delay time the DSP allows before starting to discard voice packets. The default is 160 milliseconds. |
nominal | The initial (and minimum allowed) delay time the DSP inserts before playing out voice packets. The default is 80 milliseconds |
milliseconds | Playout-delay value in milliseconds. The range for maximum playout delay is 40 to 320, and the range for nominal playout delay is 40 to 240. |
The default maximum delay is 160 milliseconds.
The default nominal delay is 80 milliseconds.
Voice-port configuration
| Release | Modification |
|---|---|
11.3 MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
If there is excessive break-up of voice due to jitter with the default playout delay settings, increase the delay times. If your network is small and jitter is minimal, decrease the delay times to reduce delay.
The following example configures a nominal playout delay of 80 milliseconds and a maximum playout delay of 160 milliseconds on voice-port 1/1 on a Cisco MC3810:
router(config)# voice-port 1/1router(config-voiceport)#playout-delay nominal 80router(config-voiceport)#playout-delay maximum 160
The following example configures a nominal playout delay of 80 milliseconds and a maximum playout delay of 160 milliseconds on voice-port 1/0/0 on the Cisco 2600 or 3600:
router(config)# voice-port 1/0/0router(config-voiceport)#playout-delay nominal 80router(config-voiceport)#playout-delay maximum 160
| Command | Description |
vad | Enables voice activity detection. |
To specify a ISDN Primary Rate interface (PRI) on a channelized T1 or E1 controller, enter the pri-group controller configuration command. Enter the no form of this command removes the remove the ISDN-PRI configuration.
pri-group timeslots timeslot-range
timeslot-range | timeslot-list is a single timeslot number, a single range of values. For T1, the allowable range is from 1 to 23. For E1, the allowable values are from 1 to 15. |
There is no ISDN-PRI group configured.
Controller configuration
| Release | Modification |
|---|---|
12.0(2)T | The command was introduced for the Cisco MC3810 multiservice access concentrator. |
12.0(7)XK | The command was introduced for the Cisco 2600 and 3600 series with a different name and some keyword modifications. |
The pri-group command applies to the configuration of Voice over Frame Relay, Voice over ATM, and Voice over HDLC on the Cisco MC3810 multiservice concentrator and the Cisco 2600 and 3600 series routers.
Before you enter the pri-group command, you must specify an ISDN-PRI switch type and an E1 or T1 controller. Only one pri group can be configured on a controller.
The following example configures configures ISDN-PRI on all timeslots of controller E1 1 on a Cisco 2600 series router::
isdn switch-type primary-qsig-master controller T1 1 pri-group timeslots 1-23
| Command | Description |
isdn switch-type | To configure the Cisco 2600 series router PRI interface to support QSIG signalling, enter this command. |
To specify the ring cadence for an FXS voice port, use the ring cadence voice-port configuration command. Use the no form of this command to restore the default value.
ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07 | pattern08 | pattern09 | pattern10 | pattern11 | pattern12] [define pulse interval]}
pattern01 | 2 seconds on, 4 seconds off |
pattern02 | 1 second on, 4 seconds off |
pattern03 | 1.5 seconds on, 3.5 seconds off |
pattern04 | 1 second on, 2 seconds off |
pattern05 | 1 second on, 5 seconds off |
pattern06 | 1 second on, 3 seconds off |
pattern07 | 0.8 second on, 3.2 seconds off |
pattern08 | 1.5 seconds on, 3 seconds off |
pattern09 | 1.2 seconds on, 3.7 seconds off |
pattern09 | 1.2 seconds on, 4.7 seconds off |
pattern11 | 0.4 second on, 0.2 second off, 0.4 second on, 2 seconds off |
pattern12 | 0.4 second on, 0.2 second off, 0.4 second on, 2.6 seconds off |
define | User-definable ring cadence pattern. Each number pair specifies one ring-pulse time and one ring-interval time. You must enter numbers in pairs, and you can enter 1 to 6 pairs. The second number in the last pair that you enter specifies the interval between rings. |
pulse | A number (1 or 2 digits) specifying ring pulse (on) time in hundreds of milliseconds. The range is 1 to 50, for pulses of 100 ms to 5000 ms. |
interval | A number (1 or 2 digits) specifying ring interval (off) time in hundreds of milliseconds. The range is 1 to 50, for pulses of 100 to 5000 ms. |
Ring cadence defaults to the pattern you specify with the cptone command.
Voice-port configuration
| Release | Modification |
|---|---|
11.3 MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers, and the patternXX syntax was introduced. |
The patternXX keyword provides preset ring cadence patterns for use on any platform. The define keyword allows you to create a custom ring cadence. On the Cisco 2600 and 3600 series routers, only one or two pairs of digits can be entered under the define keyword.
The following example configures the ring cadence for 1 second on and 4 seconds off on voice port 1/1 on a Cisco MC3810:
router(config)# voice-port 1/1
router(config-voiceport)#ring cadence pattern02
The following example configures the ring cadence for 1 second on, 1 second off, 1 second on, and 5 seconds off on voice port 1/2 on a Cisco MC3810:
voice-port 1/2
router(config-voiceport)#ring cadence define 10 10 10 50
The following example configures the ring cadence for 1 second on and 2 seconds off on voice port 1/0/0 on a Cisco 2600 or 3600:
router(config)# voice-port 1/0/0
router(config-voiceport)#ring cadence pattern04
| Command | Description |
ring frequency | Specifies the ring frequency for an FXS voice port. |
cptone | Specifies the default tone, ring, and cadence settings according to country. |
To configure a network-specific address for a dial peer, use the session target dial-peer configuration command. Use the no form of this command to disable this feature.
Cisco MC3810 Voice over IP:
session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp | loopback:compressed | loopback:uncompressed}Cisco 3600 Voice over ATM:
session target interface pvc {name | vpi/vci | vci}For the Cisco MC3810 Voice over IP:
ipv4:destination-address | IP address of the dial peer. |
dns:host-name | Indicates that the domain name server will be used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device. (Optional) You can use one of the following three wildcards with this keyword when defining the session target for VoIP peers:
|
loopback:rtp | Indicates that all voice data will be looped back to the originating source. This is applicable for VoIP peers. |
loopback:compressed | Indicates that all voice data will be looped back in compressed mode to the originating source. This is applicable for POTS peers. |
loopback:uncompressed | Indicates that all voice data will be looped-back in uncompressed mode to the originating source. This is applicable for POTS peers. |
For Cisco 3600 series Voice over ATM dial peers:
interface | Interface type and interface number on the router. |
pvc | The specific ATM permanent virtual circuit (PVC) for this dial peer. |
name | The PVC name. |
vpi/vci | ATM network virtual path identifier (VPI) and virtual channel identifier (VCI) of this PVC. On the Cisco 3600, if you have the Multiport T1/E1 ATM network module with IMA installed, the valid range for vpi is 0-15, and the valid range for vci is 1-255. If you have the OC3 ATM Network Module installed, the valid range for vpi is 0-15, and the valid range for vci is 1-1023. |
vci | ATM network virtual channel identifier (VCI) of this PVC. |
Enabled with no IP address or domain name defined.
Dial-peer configuration
| Release | Modification |
|---|---|
11.3(1) T | This command was first introduced. |
11.3(1) MA | Support was added for VoFR,VoATM and VoHDLC dial peers on the Cisco MC38110. |
12.0(3) XG and 12.0(4)T | The cid option was added. Support was added for VoFR dial peers on the Cisco 2600 and Cisco 3600 series routers. |
12.0(7)XK | Support was added for VoATM dial peers on the Cisco 3600 series routers. Support was added for VoIP dial peers on the Cisco MC3810. Support for VoHDLC on the Cisco MC3810 was removed in this release. |
This command applies to both the Cisco 3600 series and the Cisco MC3810.
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origination and the loopback type selected.
The session target dns command can be used with or without the specified wildcards. Using the optional wildcards can reduce the number of VoIP dial peer session targets you need to configure if you have groups of numbers associated with a particular router.
The following example configures a session target using DNS for a host, "voice_router," in the domain "cisco.com":
dial-peer voice 10 voip session target dns:voice_router.cisco.com
The following example configures a session target using DNS, with the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched portion of the dialed number—in this case, the four-digit extension, to identify the dial peer. As in the previous example, the domain is "cisco.com."
dial-peer voice 10 voip destination-pattern 1310222.... session target dns:$u$.cisco.com
The following example configures a session target using dns, with the optional $d$. wildcard. In this example, the destination pattern has been configured for 13102221111. The optional wildcard $d$. indicates that the router will use the destination pattern to identify the dial peer in the "cisco.com" domain.
dial-peer voice 10 voip destination-pattern 13102221111 session target dns:$d$.cisco.com
The following example configures a session target using DNS, with the optional $e$. wildcard. In this example, the destination pattern has been configured for 12345. The optional wildcard $e$. indicates that the router will reverse the digits in the destination pattern, add periods between the digits, and then use this reverse-exploded destination pattern to identify the dial peer in the "cisco.com" domain.
dial-peer voice 10 voip destination-pattern 12345 session target dns:$e$.cisco.com
The following example configures a session target for Voice over ATM on the Cisco 3600 series. The session target is sent to ATM interface 0, and is for a PVC with a VPI/VCI of 1/100.
router(config)# dial-peer voice 12 voatm router(config-dial-peer)# destination-pattern 13102221111 router(config-dial-peer)# session target atm1/0 pvc 1/100
| Command | Description |
called-number | Enables an incoming VoFR call leg to be bridged to the correct POTS call leg. |
codec (dial-peer) | Specifies the voice coder rate of speech for a dial peer. |
cptone | Specifies a regional tone, ring, and cadence setting for an analog voice port. |
destination-pattern | Specifies either the prefix or the full E.164 telephone number to be used for a dial peer. |
dtmf-relay | Enables the DSP to generate FRF.11 Annex A frames for a dial peer. |
preference | Indicates the preferred selection order of a dial peer within a hunt group. |
session protocol | Establishes a VoFR protocol for calls between the local and the remote routers via the packet network. |
To show the active call table, use the show call active voice EXEC command.
show call active voiceThis command has no arguments or keywords.
User EXEC
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced on the Cisco 2600 and 3600. |
12.0(3)XG | Support for VoFR was added. |
12.0(4)T | This command was first supported on the Cisco 7200 series. |
12.0(7)XK | This command was first supported on the Cisco MC3810 series. |
This command applies to Voice over IP, Voice over Frame Relay, and Voice over ATM on the Cisco 2600, 3600, and MC3810 series.
Use this command to display the contents of the active call table, which shows all of the calls currently connected through the router. This command displays information about call times, dial peers, connections, Quality of Service, and other status and statistical information.
See Table 3 for a listing of the information types associated with this command.
The following is sample output from the show call active voice command:
router#show call active voiceGENERIC: SetupTime=21072 Index=0 PeerAddress= PeerSubAddress= PeerId=0
PeerIfIndex=0 LogicalIfIndex=0 ConnectTime=0 CallState=3 CallOrigin=2 ChargedUnits=0
InfoType=0 TransmitPackets=375413 TransmitBytes=7508260 ReceivePackets=377734
ReceiveBytes=7554680
VOIP: ConnectionId[0x19BDF910 0xAF500007 0x0 0x58ED0] RemoteIPAddress=17635075
RemoteUDPPort=16394 RoundTripDelay=0 SelectedQoS=0 SessionProtocol=1
SessionTarget= OnTimeRvPlayout=0 GapFillWithSilence=0 GapFillWithPrediction=600
GapFillWithInterpolation=0 GapFillWithRedundancy=0 HiWaterPlayoutDelay=110
LoWaterPlayoutDelay=64 ReceiveDelay=94 VADEnable=0 CoderTypeRate=0
GENERIC: SetupTime=21072 Index=1 PeerAddress=+14085271001 PeerSubAddress=
PeerId=0 PeerIfIndex=0 LogicalIfIndex=5 ConnectTime=21115 CallState=4 CallOrigin=1
ChargedUnits=0 InfoType=1 TransmitPackets=377915 TransmitBytes=7558300
ReceivePackets=375594 ReceiveBytes=7511880
TELE: ConnectionId=[0x19BDF910 0xAF500007 0x0 0x58ED0] TxDuration=16640
VoiceTxDuration=16640 FaxTxDuration=0 CoderTypeRate=0 NoiseLevel=0 ACOMLevel=4
OutSignalLevel=-440 InSignalLevel=-440 InfoActivity=2 ERLLevel=227
SessionTarget=
Table 3 provides an alphabetical listing of the fields in this output and a description of each field.
| Field | Description |
|---|---|
ACOM Level | Current ACOM level for the call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call. |
CallOrigin | Call origin; answer versus originate. |
CallState | Current state of the call. |
CoderTypeRate | Negotiated coder transmit rate of voice/fax compression during the call. |
ConnectionId | Global call identifier of a gateway call. |
ConnectTime | Time at which the call was connected. |
Dial-Peer | Tag of the dial peer transmitting this call. |
ERLLevel | Current Echo Return Loss (ERL) level for this call. |
FaxTxDuration | Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value. |
GapFillWithSilence | Duration of voice signal replaced with silence because voice data was lost or not received on time for this call. |
GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call. An example of such pullout is frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms. |
GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received on time from voice gateway for this call. |
GapFillWithRedundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because voice data was lost or not received on time from voice gateway for this call. |
HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during this call. |
Index | Dial peer identification number. |
InfoActivity | Active information transfer activity state for this call. |
InfoType | Information type for this call. |
InSignalLevel | Active input signal level from the telephony interface used by this call. |
LogicalIfIndex | Index number of the logical interface for this call. |
LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during the call. |
NoiseLevel | Active noise level for the call. |
OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
OutSignalLevel | Active output signal level to telephony interface used by this call. |
PeerAddress | Destination pattern associated with this peer. |
PeerId | ID value of the peer table entry to which this call was made. |
PeerIfIndex | Voice port index number for this peer. |
PeerSubaddress | Subaddress to which this call is connected. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the decoder delay during the voice call. |
ReceivePackets | Number of packets received by this peer during this call. |
RemoteIPAddress | Remote system IP address for the VoIP call. |
RemoteUDPPort | Remote system UDP listener port to which voice packets are transmitted. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone during the call. |
SelectedQoS | Selected RSVP quality of service (QoS) for the call. |
SessionProtocol | Session protocol used for an Internet call between the local and remote router via the IP backbone. |
SessionTarget | Session target of the peer used for the call. |
SetupTime | Value of the System UpTime when the call associated with this entry was started. |
TransmitBytes | Number of bytes transmitted from this peer during the call. |
TransmitPackets | Number of packets transmitted from this peer during the call. |
TxDuration | Duration of transmit path open from this peer to the voice gateway for the call. |
VADEnable | Whether or not voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice transmission from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value. |
| Command | Description |
Displays the call history table. | |
show dial-peer voice | Displays configuration information for dial peers. |
show num-exp | Displays the number expansions configured. |
Displays configuration information about a specific voice port. |
To display the call history table, use the show call history voice EXEC command.
show call history voice [last number | brief]
last number | (Optional) Displays the last calls connected, where the number of calls displayed is defined by the argument number. Valid entries for the argument number are numbers from 1 to 2147483647. |
brief | (Optional) Displays abbreviated call history information for each leg of a call. |
User EXEC
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced on the Cisco 3600. |
12.0(3)XG | Support for VoFR was added. |
12.0(4)T | The brief keyword was added and the command was first supported on the Cisco 7200 series. |
12.0(7)XK | Support for brief the keyword was added on the Cisco MC3810 platform. |
This command applies to all voice applications on the Cisco 2600, 3600, MC3810, and 7200 platforms.
Use the show call history voice privileged EXEC command to display the call history table. The call history table contains a listing of all voice calls connected through this router in descending time order. You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword last, and define the number of calls to be displayed with the argument number. To display a shortened version of the call history table, use the keyword brief.
The following is sample output from the show call history voice command for a VoFR call using the frf11-trunk session protocol:
router# show call history voice last 1GENERIC: SetupTime=8283963 ms Index=3149 PeerAddress=3623110 PeerSubAddress= PeerId=3400 PeerIfIndex=18 LogicalIfIndex=0 DisconnectCause=3F DisconnectText=service or option not available, unspecified ConnectTime=8283963 DisconectTime=8285463 CallOrigin=1 ChargedUnits=0 InfoType=2 TransmitPackets=94 TransmitBytes=2751 ReceivePackets=0 ReceiveBytes=0 VOFR: ConnectionId=[0x3D4B232D 0x6A900627 0x0 0x4F00852] Subchannel=[Interface Serial0/0, DLCI 160, CID 10] SessionProtocol=frf11-trunk SessionTarget=Serial0/0 160 10 CalledNumber=2603100 VADEnable=ENABLED CoderTypeRate=g729r8 CodecBytes=30 SignalingType=cas DTMFRelay=DISABLED UseVoiceSequenceNumbers=DISABLED GENERIC: SetupTime=8283963 ms Index=3150 PeerAddress=2601100 PeerSubAddress= PeerId=1100 PeerIfIndex=7 LogicalIfIndex=0 DisconnectCause=3F DisconnectText=service or option not available, unspecified ConnectTime=8283964 DisconectTime=8285464 CallOrigin=2 ChargedUnits=0 InfoType=2 TransmitPackets=0 TransmitBytes=-121 ReceivePackets=94 ReceiveBytes=2563 TELE: ConnectionId=[0x3D4B232D 0x6A900627 0x0 0x4F00852] TxDuration=15000 ms VoiceTxDuration=2010 ms FaxTxDuration=0 ms CoderTypeRate=g729r8 NoiseLevel=-68 ACOMLevel=20 SessionTarget=
The following is sample output from the show call history voice command for a VoIP call:
router#show call history voiceGENERIC: SetupTime=20405 Index=0 PeerAddress= PeerSubAddress= PeerId=0 PeerIfIndex=0 LogicalIfIndex=0 DisconnectCause=NORMAL DisconnectText= ConnectTime=0 DisconectTime=20595 CallOrigin=2 ChargedUnits=0 InfoType=0 TransmitPackets=0 TransmitBytes=0 ReceivePackets=0 ReceiveBytes=0 VOIP: ConnectionId[0x19BDF910 0xAF500006 0x0 0x56590] RemoteIPAddress=17635075 RemoteUDPPort=16392 RoundTripDelay=0 SelectedQoS=0 SessionProtocol=1 SessionTarget= OnTimeRvPlayout=0 GapFillWithSilence=0 GapFillWithPrediction=0 GapFillWithInterpolation=0 GapFillWithRedundancy=0 HiWaterPlayoutDelay=0 LoWaterPlayoutDelay=0 ReceiveDelay=0 VADEnable=0 CoderTypeRate=0 TELE: ConnectionId=[0x19BDF910 0xAF500006 0x0 0x56590] TxDuration=3030 VoiceTxDuration=2700 FaxTxDuration=0 CoderTypeRate=0 NoiseLevel=0 ACOMLevel=0 SessionTarget=
Table 4 provides an alphabetical listing of the fields in this output and a description of each field.
| Field | Description |
|---|---|
ACOMLevel | Average ACOM level for this call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call. |
CallOrigin | Call origin; answer versus originate. |
CoderTypeRate | Negotiated coder rate. This value specifies the transmit rate of voice/fax compression to its associated call leg for the call. |
ConnectionID | Global call identifier for the gateway call. |
ConnectTime | Time the call was connected. |
DisconnectCause | Description explaining why the call was disconnected. |
DisconnectText | Descriptive text explaining the disconnect reason. |
DisconnectTime | Time the call was disconnected. |
FaxDuration | Duration of fax transmitted from this peer to the voice gateway for this call. You can derive the Fax Utilization Rate by dividing this value by the TxDuration value. |
GapFillWithSilence | Duration of voice signal replaced with silence because the voice data was lost or not received on time for this call. |
GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call. |
GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call. |
GapFillWithRedundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because the voice data was lost or not received on time from the voice gateway for this call. |
HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during the voice call. |
Index | Index number identifying the voice-peer for this call. |
InfoType | Information type for this call. |
LogicalIfIndex | Index of the logical voice port for this call. |
LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during the voice call. |
NoiseLevel | Average noise level for this call. |
OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
PeerAddress | Destination pattern or number to which this call is connected. |
PeerId | ID value of the peer entry table to which this call was made. |
PeerIfIndex | Index number of the logical interface through which this call was made. For ISDN media, this would be the index number of the B channel used for the call. |
PeerSubAddress | Subaddress to which this call is connected. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the decoder delay during the voice call. |
ReceivePackets | Number of packets received by this peer during the call. |
RemoteIPAddress | Remote system IP address for the call. |
RemoteUDPPort | Remote system UDP listener port to which voice packets for this call are transmitted. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone for this call. |
SelectedQoS | Selected RSVP quality of service for the call. |
SessionProtocol | Session protocol to be used for an Internet call between the local and remote router via the IP backbone. |
SessionTarget | Session target of the peer used for the call. |
SetUpTime | Value of the System UpTime when the call associated with this entry was started. |
TransmitBytes | Number of bytes transmitted by this peer during the call. |
TransmitPackets | Number of packets transmitted by this peer during the call. |
TxDuration | Duration of the transmit path open from this peer to the voice gateway for the call. |
VADEnable | Whether or not voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice transmitted from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration by the TxDuration value. |
| Command | Description |
Displays the contents of the active call table. | |
show dial-peer voice | Displays configuration information for dial peers. |
show num-exp | Displays the number expansions configured. |
Displays configuration information about a specific voice port. |
To show the number expansions configured, use the show num-exp privileged EXEC command.
show num-exp [dialed-number]
dialed-number | (Optional) Dialed number. |
User EXEC and Privileged EXEC
| Release | Modification |
|---|---|
11.3(1)T | This command was first introduced on the Cisco 3600 platform. |
12.0(3)T | This command was first supported on the Cisco AS5300 platform. |
12.0(4)XL | This command was first supported on the Cisco AS5800 platform. |
12.0(7)XK | This command was first supported on the Cisco MC3810 platform. |
This command applies to VoFR, VoATM, and Voice over IP on the Cisco 2600 series, 3600 series, and MC3810 platforms.
Use the show num-exp privileged EXEC command to display all of the number expansions configured for this router. To display number expansion for only one number, specify that number by using the dialed-number argument.
The following is sample output from the show num-exp command:
router# show num-exp Dest Digit Pattern = '0...' Translation = '+14085270...' Dest Digit Pattern = '1...' Translation = '+14085271...' Dest Digit Pattern = '3..' Translation = '+140852703..' Dest Digit Pattern = '4..' Translation = '+140852804..' Dest Digit Pattern = '5..' Translation = '+140852805..' Dest Digit Pattern = '6....' Translation = '+1408526....' Dest Digit Pattern = '7....' Translation = '+1408527....' Dest Digit Pattern = '8...' Translation = '+14085288...'
Table 5 explains the fields in the sample output.
| Field | Description |
|---|---|
Dest Digit Pattern | Index number identifying the destination telephone number digit pattern. |
Translation | Expanded destination telephone number digit pattern. |
| Command | Description |
Displays the contents of the active call table. | |
Displays the call history table. | |
show dial-peer voice | Displays configuration information for dial peers. |
show voice port | Displays configuration information about a specific voice port. |
To show the call status for voice ports on the Cisco router or concentrator, use the show voice call EXEC command.
For the Cisco 2600 and 3600 series with analog voice ports:
show voice call [slot/subunit/port | summary]For the Cisco 2600 and 3600 series with digital voice ports (with T1 packet voice trunk network modules):
show voice call [slot/port:ds0-group | summary]For the Cisco MC3810 series with analog voice ports:
show voice call [slot/port | summary]For the Cisco MC3810 series with digital voice ports:
show voice call [slot:ds0-group | summary]
summary | (Optional) Show a summary of the call status, not the detailed report. |
voice-port | (Optional) Displays the call status for a specified voice port. |
User EXEC
| Release | Modification |
|---|---|
11.3 MA | This command was introduced for the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
This command applies to Voice over Frame Relay, Voice over ATM, and Voice over IP.
This command shows call-processing and protocol state-machine information for a voice port, if it is available. It also shows information on the DSP channel associated with the voice port, if it is available. All real-time information in the DSP channel, such as jitter and buffer overrun for example, is queried to the DSP channel, and asynchronous responses are returned to the host side.
If no call is active on a voice port, the show voice call summary command displays only the VPM (shutdown) state. If a call is active on a voice port, the VTSPS state is shown. For an on-net call or a local call without local-bypass (not cross-connected), the CODEC and VAD fields are displayed. For an off-net call or a local call with local-bypass, the CODEC and VAD fields are not displayed.
CODEC and VAD are not displayed in the show voice call port command, because this information is in the summary display.
This command provides the status at these levels of the call handling module:
The following is a sample display from the show voice call summary command for voice ports on a Cisco MC3810, showing two local calls connected without local bypass:
router# show voice call summary PORT CODEC VAD VTSP STATE VPM STATE ======= ======== === ===================== ======================== 0:17.18 *shutdown* 0:18.19 g729ar8 n S_CONNECT FXOLS_OFFHOOK 0:19.20 FXOLS_ONHOOK 0:20.21 FXOLS_ONHOOK 0:21.22 FXOLS_ONHOOK 0:22.23 FXOLS_ONHOOK 0:23.24 EM_ONHOOK 1/1 FXSLS_ONHOOK 1/2 FXSLS_ONHOOK 1/3 EM_ONHOOK 1/4 EM_ONHOOK 1/5 FXOLS_ONHOOK 1/6 g729ar8 n S_CONNECT FXOLS_CONNECT
The following is a sample display from the show voice call summary command for voice ports on a Cisco MC3810, showing two local calls connected with local bypass:
router# show voice call summary PORT CODEC VAD VTSP STATE VPM STATE ======= ======== === ===================== ======================== 0:17.18 *shutdown* 0:18.19 S_CONNECT FXOLS_OFFHOOK 0:19.20 FXOLS_ONHOOK 0:20.21 FXOLS_ONHOOK 0:21.22 FXOLS_ONHOOK 0:22.23 FXOLS_ONHOOK 0:23.24 EM_ONHOOK 1/1 FXSLS_ONHOOK 1/2 FXSLS_ONHOOK 1/3 EM_ONHOOK 1/4 EM_ONHOOK 1/5 FXOLS_ONHOOK 1/6 S_CONNECT FXOLS_CONNECT
The following is a sample display from the show voice call command for analog voice ports on a Cisco MC3810:
router# show voice call 1/1 vpm level 1 state = FXSLS_ONHOOK vpm level 0 state = S_UP 1/2 vpm level 1 state = FXSLS_ONHOOK vpm level 0 state = S_UP 1/3 is shutdown 1/4 vtsp level 0 state = S_CONNECT vpm level 1 state = S_TRUNKED vpm level 0 state = S_UP 1/5 vpm level 1 state = EM_ONHOOK vpm level 0 state = S_UP 1/6 vpm level 1 state = EM_ONHOOK vpm level 0 state = S_UP sys252#show voice call 1/4 1/4 vtsp level 0 state = S_CONNECT vpm level 1 state = S_TRUNKED vpm level 0 state = S_UP router# ***DSP VOICE VP_DELAY STATISTICS*** Clk Offset(ms): 1445779863, Rx Delay Est(ms): 95 Rx Delay Lo Water Mark(ms): 95, Rx Delay Hi Water Mark(ms): 125 ***DSP VOICE VP_ERROR STATISTICS*** Predict Conceal(ms): 10, Interpolate Conceal(ms): 0 Silence Conceal(ms): 0, Retroact Mem Update(ms): 0 Buf Overflow Discard(ms): 20, Talkspurt Endpoint Detect Err: 0 ***DSP VOICE RX STATISTICS*** Rx Vox/Fax Pkts: 537, Rx Signal Pkts: 0, Rx Comfort Pkts: 0 Rx Dur(ms): 50304730, Rx Vox Dur(ms): 16090, Rx Fax Dur(ms): 0 Rx Non-seq Pkts: 0, Rx Bad Hdr Pkts: 0 Rx Early Pkts: 0, Rx Late Pkts: 0 ***DSP VOICE TX STATISTICS*** Tx Vox/Fax Pkts: 567, Tx Sig Pkts: 0, Tx Comfort Pkts: 0 Tx Dur(ms): 50304730, Tx Vox Dur(ms): 17010, Tx Fax Dur(ms): 0 ***DSP VOICE ERROR STATISTICS*** Rx Pkt Drops(Invalid Header): 0, Tx Pkt Drops(HPI SAM Overflow): 0 ***DSP LEVELS*** TDM Bus Levels(dBm0): Rx -70.3 from PBX/Phone, Tx -68.0 to PBX/Phone TDM ACOM Levels(dBm0): +2.0, TDM ERL Level(dBm0): +5.6 TDM Bgd Levels(dBm0): -71.4, with activity being voice
| Command | Description |
show dial-peer voice | Displays the configuration for all VoIP and POTS dial peers configured on the router. |
show voice dsp | Shows the current status of all DSP voice channels. |
Displays configuration information about a specific voice port. |
To show the configuration status for all configured DSP voice channels on the Cisco router or concentrator, use the show voice dsp EXEC command.
show voice dspThis command has no arguments or keywords.
User EXEC
| Release | Modification |
|---|---|
11.3 MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600, and the display format was modified. |
This command applies to Voice over Frame Relay, Voice over ATM, and Voice over IP.
Use this command when abnormal behavior in the DSP voice channels occurs.
The following is a sample display from the show voice dsp command on a Cisco MC3810:
Router#show voice dsp
BOOT PAK
TYPE DSP CH CODEC VERS STATE STATE RST AI PORT TS ABORT TX/RX-PAK-CNT
==== === == ======== ==== ===== ======= === == ======= == ===== ===============
C549 001 01 {high} 3.3 idle idle 6 0 0 1365/1364
02 {high} idle 0 0/0
C549 002 01 {high} 3.3 idle idle 6 0 0 1365/1364
02 {high} idle 0 0/0
C549 003 01 {high} 3.3 idle idle 6 0 0 1365/1364
02 {high} idle 0 0/0
C549 004 01 {high} 3.3 idle idle 6 0 0 1365/1364
02 {high} idle 0 0/0
C549 005 01 {high} 3.3 idle idle 6 0 0 1365/1364
02 {high} idle 0 0/0
C549 006 01 {high} 3.3 idle idle 6 0 0 1365/1364
02 {high} idle 0 0/0
Table 6 provides an alphabetical listing of the fields in this output and a description of each field.
| Field | Description |
|---|---|
AI | Number of alarm indications received from the DSP, which may point to abnormality of DSP firmware. |
BOOT STATE | Applicable to Cisco MC3810 only of dynamic reload of DSP is permitted. |
CH | Voice channel number in DSP. |
CODEC | Cisco MC3810 with HCM and Cisco 2600 and 3600 digital:
Cisco MC3810 with VCM and Cisco 2600 and 3600 analog:
|
DSP | DSP number. |
PAK ABORT | The number of DSP packets dropped due to DSP failure in picking up packets from the host. |
PORT | The port number associated with the DSP channel. This is a fixed port number on the Cisco 2600 and 3600; this number may change with each new call on the Cisco MC3810. |
RST | The number of DSP resets since the most recent clear counters entry. |
STATE | The busy/idle state of the DSP channel. |
TS | The backplane timeslot associated with this DSP channel. This is a fixed timeslot on the Cisco 2600 and 3600; this number may change with each new call on the Cisco MC3810. |
TX/RX-PAK-CNT | An ordered pair of transmit and receive packet counts processed by the DSP since the previous clear counters command was entered. |
TYPE | DSP hardware type. |
VERS | Version and revision of DSP hardware, in X,Y format. |
| Command | Description |
clear counters | Clears all the current interface counters from the interface. |
Displays configuration information about a specific voice port. |
To display configuration information about a specific voice port, use the show voice port EXEC command.
For the Cisco 2600 and 3600 series with analog voice ports:
show voice port [slot/subunit/port | summary]For the Cisco 2600 and 3600 series with digital voice ports (with T1 packet voice trunk network modules):
show voice port [slot/port:ds0-group | summary]For the Cisco MC3810 series with analog voice ports:
show voice port [slot/port | summary]For the Cisco MC3810 series with digital voice ports:
show voice port [slot:ds0-group | summary]For the Cisco 2600 and 3600 series with analog voice ports:
slot/subunit/port | (Optional) Displays information for the analog voice port you specify with the slot/subunit/port designation. slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform. subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. (The VIC fits into the voice network module.) port specifies an analog voice port number. Valid entries are 0 and 1. |
summary | (Optional) Displays a summary of all voice ports. |
For the Cisco 2600 and 3600 series with digital voice ports:
slot/port:ds0-group | (Optional) Displays information for the digital voice port you specify with the slot/port:ds0-group designation. slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform. port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. (One VWIC fits in an NM.) ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
summary | (Optional) Displays a summary of all voice ports. |
For the Cisco MC3810 series with analog voice ports:
slot/port | (Optional) Displays information for the analog voice port you specify with the slot/port designation. slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810. port specifies an analog voice port number. Valid entries are 1 to 6. |
summary | (Optional) Displays a summary of all voice ports. |
For the Cisco MC3810 series with digital voice ports:
slot:ds0-group | (Optional) Displays information for the digital voice port you specify with the slot:ds0-group designation. slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1). ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
summary | (Optional) Displays a summary of all voice ports. |
User EXEC
| Release | Modification |
|---|---|
11.3(1) T | This command was introduced. |
12.0(5)XK and 12.0(7)T | The ds0-group argument was added for the Cisco 2600 and 3600 series routers. |
12.0(7)XK | The summary keyword was added for the Cisco 2600 and 3600 series routers. The ds0-group argument was added for the Cisco MC3810. |
Use the show voice port privileged EXEC command to display configuration and voice-interface-card-specific information about a specific port.
The following is sample output from the show voice port summary command for all voice ports on a Cisco MC3810 with an analog voice module (AVM):
router# show voice port summary IN OUT ECHO PORT SIG-TYPE ADMIN OPER IN-STATUS OUT-STATUS GAIN ATTN CANCEL 1/1 fxs-ls up up on-hook idle 0 0 y 1/2 fxs-ls up up on-hook idle 0 0 y 1/3 e&m-wnk up up idle idle 0 0 y 1/4 e&m-wnk up up idle idle 0 0 y 1/5 fxo-ls up up idle on-hook 0 0 y 1/6 fxo-ls up up idle on-hook 0 0 y
The following is sample output from the show voice port summary command on a Cisco MC3810 with a digital voice module (DVM):
IN OUT PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC ====== == ========== ===== ==== ======== ======== == 0:17 18 fxo-ls down down idle on-hook y 0:18 19 fxo-ls up dorm idle on-hook y 0:19 20 fxo-ls up dorm idle on-hook y 0:20 21 fxo-ls up dorm idle on-hook y 0:21 22 fxo-ls up dorm idle on-hook y 0:22 23 fxo-ls up dorm idle on-hook y 0:23 24 e&m-imd up dorm idle idle y 1/1 -- fxs-ls up dorm on-hook idle y 1/2 -- fxs-ls up dorm on-hook idle y 1/3 -- e&m-imd up dorm idle idle y 1/4 -- e&m-imd up dorm idle idle y 1/5 -- fxo-ls up dorm idle on-hook y 1/6 -- fxo-ls up dorm idle on-hook y Elements : sys/voip/ccvpm vpm_htsp.c (107) sys/voip/ccvtsp vtsp_core.c (167) sys/voip/cli voiceport_action.c (58)
The following is sample output from the show voice port command for an E&M analog voice port on a Cisco 3600:
router#show voice port 1/0/0E&M Slot is 1, Sub-unit is 0, Port is 0Type of VoicePort is E&MOperation State is unknownAdministrative State is unknownThe Interface Down Failure Cause is 0Alias is NULLNoise Regeneration is disabledNon Linear Processing is disabledMusic On Hold Threshold is Set to 0 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is disabledEcho Cancel Coverage is set to 16msConnection Mode is NormalConnection Number isInitial Time Out is set to 0 sInterdigit Time Out is set to 0 sAnalog Info Follows:Region Tone is set for northamericaCurrently processing noneMaintenance Mode Set to None (not in mtc mode)Number of signaling protocol errors are 0Voice card specific Info Follows:Signal Type is wink-startOperation Type is 2-wireImpedance is set to 600r OhmE&M Type is unknownDial Type is dtmfIn Seizure is inactiveOut Seizure is inactiveDigit Duration Timing is set to 0 msInterDigit Duration Timing is set to 0 msPulse Rate Timing is set to 0 pulses/secondInterDigit Pulse Duration Timing is set to 0 msClear Wait Duration Timing is set to 0 msWink Wait Duration Timing is set to 0 msWink Duration Timing is set to 0 msDelay Start Timing is set to 0 msDelay Duration Timing is set to 0 ms
The following is sample output from the show voice port command for an FXS analog voice port on a Cisco 3600:
router# show voice port 1/0/0 Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0, Port is 0 Type of VoicePort is FXS Operation State is DORMANT Administrative State is UP The Interface Down Failure Cause is 0 Alias is NULL Noise Regeneration is enabled Non Linear Processing is enabled Music On Hold Threshold is Set to 0 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 16ms Connection Mode is Normal Connection Number is Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Analog Info Follows: Region Tone is set for northamerica Currently processing none Maintenance Mode Set to None (not in mtc mode) Number of signaling protocol errors are 0 Voice card specific Info Follows: Signal Type is loopStart Ring Frequency is 25 Hz Hook Status is On Hook Ring Active Status is inactive Ring Ground Status is inactive Tip Ground Status is inactive Digit Duration Timing is set to 100 ms InterDigit Duration Timing is set to 100 ms Hook Flash Duration Timing is set to 600 ms
The following is sample output from the show voice port command for an FXS analog voice port on a Cisco MC3810:
router# show voice port 1/2
Voice port 1/2 Slot is 1, Port is 2
Type of VoicePort is FXS
Operation State is UP
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Coder Type is g729ar8
Companding Type is u-law
Voice Activity Detection is disabled
Ringing Time Out is 180 s
Wait Release Time Out is 30 s
Nominal Playout Delay is 80 milliseconds
Maximum Playout Delay is 160 milliseconds
Analog Info Follows:
Region Tone is set for northamerica
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Analog interface A-D gain offset = -3 dB
Analog interface D-A gain offset = -3 dB
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 20 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is active
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Ring Cadence are [20 40] * 100 msec
InterDigit Pulse Duration Timing is set to 500 ms
The following is sample output from the show voice port command for an E&M digital voice port on a Cisco 3600:
router# show voice port 1/0:1 receEive and transMit Slot is 1, Sub-unit is 0, Port is 1 Type of VoicePort is E&M Operation State is DORMANT Administrative State is UP No Interface Down Failure Description is not set Noise Regeneration is enabled Non Linear Processing is enabled Music On Hold Threshold is Set to -38 dBm In Gain is Set to 0 dB Out Attenuation is Set to 0 dB Echo Cancellation is enabled Echo Cancel Coverage is set to 8 ms Connection Mode is normal Connection Number is not set Initial Time Out is set to 10 s Interdigit Time Out is set to 10 s Region Tone is set for US
Table 7 explains the fields in the sample output.
| Field | Description |
|---|---|
Administrative State | Administrative state of the voice port. |
Alias | User-supplied alias for this voice port. |
Analog interface A-D gain offset | Offset of the gain for analog-to-digital conversion. |
Analog interface D-A gain offset | Offset of the gain for digital-to-analog conversion. |
Clear Wait Duration Timing | Time of inactive seizure signal to declare call cleared. |
Coder Type | Voice compression mode used. |
Companding Type | Companding standard used to convert between analog and digital signals in PCM systems. |
Connection Mode | Connection mode of the interface. |
Connection Number | Full E.164 telephone number used to establish a connection with the trunk or PLAR mode. |
Currently Processing | Type of call currently being processed: none, voice, or fax. |
Delay Duration Timing | Maximum delay signal duration for delay dial signaling. |
Delay Start Timing | Timing of generation of delayed start signal from detection of incoming seizure. |
Description | Description of the voice port. |
Dial Type | Out-dialing type of the voice port. |
Digit Duration Timing | DTMF Digit duration in milliseconds. |
E&M Type | Type of E&M interface. |
Echo Cancel Coverage | Echo Cancel Coverage for this port. |
Echo Cancellation | Whether or not echo cancellation is enabled for this port. |
Hook Flash Duration Timing | Maximum length of hook flash signal. |
Hook Status | Hook status of the FXO/FXS interface. |
Impedance | Configured terminating impedance for the E&M interface. |
In Gain | Amount of gain inserted at the receiver side of the interface. |
In Seizure | Incoming seizure state of the E&M interface. |
Initial Time Out | Amount of time the system waits for an initial input digit from the caller. |
InterDigit Duration Timing | DTMF interdigit duration in milliseconds. |
InterDigit Pulse Duration Timing | Pulse dialing interdigit timing in milliseconds. |
Interdigit Time Out | Amount of time the system waits for a subsequent input digit from the caller. |
Maintenance Mode | Maintenance mode of the voice port. |
Maximum Playout Delay | The amount of time before the Cisco MC3810 DSP starts to discard voice packets from the DSP buffer. |
Music On Hold Threshold | Configured Music-On-Hold Threshold value for this interface. |
Noise Regeneration | Whether or not background noise should be played to fill silent gaps if VAD is activated. |
Nominal Playout Delay | The amount of time the Cisco MC3810 DSP waits before starting to play out the voice packets from the DSP buffer. |
Non-Linear Processing | Whether or not non-linear processing is enabled for this port. |
Number of signaling protocol errors | Number of signaling protocol errors. |
Operations State | Operation state of the port. |
Operation Type | Operation of the E&M signal: two-wire or four-wire. |
Out Attenuation | Amount of attenuation inserted at the transmit side of the interface. |
Out Seizure | Outgoing seizure state of the E&M interface. |
Port | Port number for this interface associated with the voice interface card. |
Pulse Rate Timing | Pulse dialing rate in pulses per second (pps). |
Region Tone | Configured regional tone for this interface. |
Ring Active Status | Ring active indication. |
Ring Cadence | Configured ring cadence for this interface. |
Ring Frequency | Configured ring frequency for this interface. |
Ring Ground Status | Ring ground indication. |
Ringing Time Out | Ringing time out duration. |
Signal Type | Type of signaling for a voice port: loop-start, ground-start, wink-start, immediate, and delay-dial. |
Slot | Slot used in the voice interface card for this port. |
Sub-unit | Subunit used in the voice interface card for this port. |
Tip Ground Status | Tip ground indication. |
Type of VoicePort | Type of voice port: FXO, FXS, and E&M. |
The Interface Down Failure Cause | Text string describing why the interface is down, |
Voice Activity Detection | Whether Voice Activity Detection is enabled or disabled. |
Wait Release Time Out | The time a voice port stays in the call-failure state while the Cisco MC3810 sends a busy tone, reorder tone, or an out-of-service tone to the port. |
Wink Duration Timing | Maximum wink duration for wink start signaling. |
Wink Wait Duration Timing | Maximum wink wait duration for wink start signaling. |
| Command | Description |
Displays the call status for all voice ports on the Cisco router or concentrator. | |
Displays the call history table. | |
show dial-peer voice | Displays configuration information about dial peers. |
show num-exp | Displays the number expansions that are configured. |
To display the status of trunk-conditioning signaling and timing parameters for a voice port, use the show voice trunk-conditioning signaling EXEC command.
show voice trunk-conditioning signaling [summary | voice-port]
summary | (Optional) Show a summary of the status for all voice ports on the router or concentrator. |
voice-port | (Optional) Show a detailed report for a specified voice port. |
EXEC
| Release | Modification |
|---|---|
12.0(3)XG and 12.0(4)T | This command was introduced on the Cisco MC3810 as show voice permanent-call. |
12.0(7)XK | This command was renamed show voice trunk-conditioning signaling. |
This command displays the trunk signaling status for analog and digital voice ports on Cisco MC3810 concentrators.
The following is a sample display from the show voice trunk-conditioning signaling summary command for voice ports on a Cisco MC3810:
router# show voice trunk-conditioning signaling summary 1/1 is shutdown 1/4 is shutdown 1/5 : TX INFO :slow-mode seq#= 25, sig pkt cnt= 40, last-ABCD=0000 hardware-state ACTIVE signal type is NorthamericanCAS signal path is OPEN RX INFO :slow-mode, sig pkt cnt= 36, prev-seq#= 25, last-ABCD=0000
The following is a sample display from the show voice trunk-conditioning signaling command for voice port 1/5 on a Cisco MC3810:
router# show voice trunk-conditioning signaling 1/5 1/5 : TX INFO :slow-mode seq#= 25, sig pkt cnt= 42, last-ABCD=0000 hardware-state ACTIVE signal type is NorthamericanCAS signal path is OPEN 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 RX INFO :slow-mode, sig pkt cnt= 37 missing = 0, out of seq = 0, very late = 0 playout depth = 0 (ms), refill count = 1 prev-seq#= 25, last-ABCD=0000 trunk_down_timer = 4212 (ms), idle timer = 0 (sec), tx_oos_timer = 0 (sec), rx_ais_duration = 0 (ms) forced playout signal pattern = NONE signaling playout history 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000 0000
The following is a sample display from the show voice trunk-conditioning signaling summary command for voice ports on a Cisco 3600:
router# show voice trunk-conditioning signaling summary 2/0/0 is shutdown 2/0/1 is shutdown 3/0:0 8 is shutdown 3/0:1 1 is shutdown 3/0:2 2 is shutdown 3/0:3 3 is shutdown 3/0:5 5 is shutdown 3/0:6(6) : status : 3/0:7 7 is shutdown 3/1:0 8 is shutdown 3/1:1 1 is shutdown 3/1:3 3 is shutdown 3/1:5 5 is shutdown 3/1:7 7 is shutdown
The following is a sample display from the show voice trunk-conditioning signaling command for voice port 3/0:6 on a Cisco 3600:
router# show voice trunk-conditioning signaling 3/0:6 hardware-state ACTIVE signal type is NorthamericanCAS status : forced playout pattern = STOPPED trunk_down_timer = 0, rx_ais_duration = 0, idle_timer = 0
Table 8 explains the fields in the sample output.
| Field | Description |
|---|---|
current timer | Time since last signaling packets were received. |
forced playout pattern | Which forced playout pattern is sent to PBX:
|
hardware-state | Hardware state based on received IDLE pattern:
|
signal type | Signaling type used by lower level driver: Northamerica, MELCAS, transparent, or external. |
idle timer | Time the hardware on both sides has been in idle state. |
last-ABCD | Last received or transmitted signal bit pattern. |
max inter-arrival time | Maximum interval between received signaling packets. |
missing | Number of missed signal packets. |
mode | Signaling packet generation frequency:
|
out of seq | Number of out-of-sequence signal packets. |
playout depth | Number of packets in playout buffer. |
prev-seq# | Sequence number of previous signaling packet. |
refill count | Number of packets created to maintain nominal length of playout packet buffer. |
rx_ais_duration | Time since receipt of AIS indicator. |
seq# | Sequence number of signaling packet. |
sig pkt cnt | Number of transmitted or received signaling packets. |
signal path | Status of signaling path. |
signaling playout history | Signaling bits received in last 60 milliseconds. |
trunk_down_timer | Time since last signaling packets were received. |
tx_oos_timer | Time since PBX started sending OOS signaling pattern defined by signal pattern oos transmit. |
very late | Number of very late signaling packets. |
| Command | Description |
show dial-peer voice | Displays the configuration for all VoIP and POTS dial peers configured on the router. |
show voice dsp | Shows the current status of all DSP voice channels. |
show voice port | Displays configuration information about a specific voice port. |
show voice trunk-conditioning supervisory | Displays the status of trunk supervision and configuration parameters for voice ports. |
To display the status of trunk supervision and configuration parameters for voice ports, use the show voice trunk-conditioning supervisory EXEC command.
show voice trunk-conditioning supervisory [summary | voice-port]
summary | (Optional) Show a summary of the status for all voice ports on the router or concentrator. |
voice-port | (Optional) Show a detailed report for a specified voice port. |
EXEC
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced on the Cisco MC3810. |
This command displays the trunk supervision and configuration status for analog and digital voice ports.
The following is a sample display from the show voice trunk-conditioning supervisory summary command for voice ports on a Cisco MC3810:
router# show voice trunk-conditioning supervisory summary 1/1 is shutdown 1/4 is shutdown 1/5 : state : TRUNK_SC_CONNECT, voice : on , signal : on ,slave
The following is a sample display from the show voice trunk-conditioning supervisory command for voice port 1/5 on a Cisco MC3810:
router# show voice trunk-conditioning supervisory 1/5 1/5 : state : TRUNK_SC_CONNECT, voice : on, signal : on, slave status: trunk connected sequence oos : idle and oos pattern :rx_idle = 0x0 rx_oos = 0xF tx_oos = 0xF timing : idle = 0, restart = 0, standby = 0, timeout = 40 supp_all = 50, supp_voice = 0, keep_alive = 5 timer: oos_ais_timer = 0, timer = 0
The following is a sample display from the show voice trunk-conditioning supervisory summary command for voice ports on a Cisco 3600:
router# show voice trunk-conditioning supervisory summary 2/0/0 is shutdown 2/0/1 is shutdown 3/0:0 8 is shutdown 3/0:1 1 is shutdown 3/0:2 2 is shutdown 3/0:3 3 is shutdown 3/0:5 5 is shutdown 3/0:6(6) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/0:7(7) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/1:0(8) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/1:1(1) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/1:3(3) : state : TRUNK_SC_CONNECT, voice : on , signal : on ,master 3/1:5(5) is shutdown 3/1:7(7) is shutdown
The following is a sample display from the show voice trunk-conditioning supervisory command for voice port 3/0:6 on a Cisco 3600:
router# show voice trunk-conditioning supervisory 3/0:6 3/0:6(6) : state : TRUNK_SC_CONNECT, voice : on, signal : on, master status: trunk connected sequence oos : idle and oos pattern :rx_idle = 0x0 rx_oos = 0xF timing : idle = 0, restart = 0, standby = 0, timeout = 40 supp_all = 0, supp_voice = 0, keep_alive = 5 timer: oos_ais_timer = 0, timer = 0
Table 9 explains the fields in the sample output.
| Field | Description |
|---|---|
keep_alive | Signaling packets periodically sent to the far end, even if there is no signal change. These signaling packets function as keepalive messages. |
master | The voice port configured as connection trunk xxxx. |
slave | The voice port configured as connection trunk xxxx answer-mode. |
oos_ais_timer | Time since the signaling packet with AIS indicator was received. |
pattern | 4-bit signaling pattern. |
restart | The restart timeout after far end is OOS. |
rx-idle | The signaling bit pattern indicating that the far end is idle. |
rx-oos | The signaling bit pattern sent to the PBX indicating that the network is OOS. |
standby | The time before the slave side goes back to standby after far end goes OOS. |
supp_all | The timeout before suppressing transmission of voice and signaling packets to the far end after detection of PBX OOS. |
supp_voice | The timeout before suppressing transmission of voice packet to the far end after detection of PBX OOS. |
timeout | The timeout for non-receipt of keepalive packets before the far end is considered to be OOS. |
TRUNK_SC_CONNECT | Trunk conditioning supervisory component status. |
| Command | Description |
show dial-peer voice | Displays the configuration for all dial peers configured on the router. |
show voice dsp | Shows the current status of all DSP voice channels. |
show voice port | Displays configuration information about a specific voice port. |
show voice trunk-conditioning signaling | Displays the status of trunk-conditioning signaling and timing parameters for a voice port. |
To configure the ABCD signaling bit pattern for Cisco trunks and FRF.11 trunks, use the signal pattern voice-class configuration command. Use the no form of this command to restore the default.
signal pattern {idle receive | idle transmit | oos receive | oos transmit} bit-pattern
idle receive | Defines the signaling pattern for identifying an "idle" message from the network. and Defines the idle signaling pattern to be sent to the PBX if the network trunk is out of service and signal sequence oos idle-only or signal sequence oos both is configured. |
idle transmit | Defines the signaling pattern for identifying an "idle" message from the PBX. |
oos receive | Defines the OOS signaling pattern to be sent to the PBX if the network trunk is out of service and signal sequence oos oos-only or signal sequence oos both is configured. |
oos transmit | Defines the signaling pattern for identifying an OOS message from the PBX. |
bit-pattern | The ABCD signaling bit pattern. Values are 0000 to 1111. |
idle receive | For near-end E&M—0000 (for T1) or 0001 (for E1) For near-end FXO loop start—0101 For near-end FXO ground start—1111 For near-end FXS—0101 For near-end MELCAS—1101 |
idle transmit | For near-end E&M—0000 For near-end FXO—0101 For near-end FXS loop start—0101 For near-end FXS ground start—1111 For near-end MELCAS—1101 |
oos receive | For near-end E&M—1111 For near-end FXO loop start—1111 For near-end FXO ground start—0000 For near-end FXS loop start—1111 For near-end FXS ground start—0101 For near-end MELCAS—1111 |
oos transmit | No default signaling pattern is defined. |
Voice-class configuration
| Release | Modification |
|---|---|
12.0(3)XG and 12.0(4)T | This command was introduced on the Cisco MC3810. |
12.0(7)XK | Default signaling patterns were defined. |
This command defines the signaling patterns that are used to identify the idle and OOS states.
Before configuring the signaling pattern, 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. After you finish defining the voice class, you will assign it to a dial peer.
To suppress voice packets whenever the transmit or receive trunk is in the idle state, use the idle receive and idle transmit commands in conjunction with the signal timing idle suppress-voice command.
To define the signaling bit patterns to be sent to the PBX when the trunk is out of service, use the idle receive and oos receive commands.
The oos receive pattern is the pattern sent to the PBX to indicate that the network trunk is out of service. The oos receive pattern is not used for pattern matching against the signaling packets received from the network.
To "busy out" a PBX if the network connection fails, set the oos receive pattern to match the seized state (busy); then set the signal timing oos timeout value. When the timeout expires and no signaling packets have been received, the router will send the idle receive and/or oos receive pattern to the PBX, depending on which pattern is specified by the signal sequence oos command.
Use the busy seized pattern only if the PBX does not have a pattern specifically intended to indicate an OOS state. If the PBX has a specific OOS pattern, use that pattern instead.
The following example restores default signaling bit patterns for the receive and transmit idle states:
router(config)#voice class permanent 10
router(config-class)# signal keepalive 3
router(config-class)# signal timing idle suppress-voice
router(config-class)# no signal pattern idle receive
router(config-class)# no signal pattern idle transmit
router(config-class)# exit router(config)# dial-peer voice 100 vofr router(config-dial-peer)# voice-class permanent 10
The following example configures non-default signaling bit patterns for the receive and transmit idle states:
router(config)#voice class permanent 10
router(config-class)# signal keepalive 3
router(config-class)# signal timing idle suppress-voice
router(config-class)# signal pattern idle receive 0101
router(config-class)# signal pattern idle transmit 0101
router(config-class)# exit router(config)# dial-peer voice 100 vofr router(config-dial-peer)# voice-class permanent 10
The following example restores default signaling bit patterns for the receive and transmit out-of-service states:
router(config)#voice class permanent 10
router(config-class)# signal keepalive 3
router(config-class)# signal timing idle suppress-voice
router(config-class)# no signal pattern oos receive
router(config-class)# no signal pattern oos transmit router(config-class)# exit router(config)# dial-peer voice 100 vofr router(config-dial-peer)# voice-class permanent 10
The following example configures non-default signaling bit patterns for the receive and transmit out-of-service states:
router(config)#voice class permanent 10
router(config-class)# signal keepalive 3
router(config-class)# signal pattern oos receive 0001
router(config-class)# signal pattern oos transmit 0001 router(config-class)# exit router(config)# dial-peer voice 100 vofr router(config-dial-peer)# voice-class permanent 10
| Command | Description |
dial-peer voice | Enters dial-peer configuration mode and specifies a dial-peer type. |
signal keepalive | Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks. |
Specifies which signaling pattern is sent to the PBX when the far-end keepalive message is lost or AIS is received from the far end. | |
Specifies the length of time before the router stops sending voice packets after a trunk goes into the idle state. | |
signal timing oos restart | Specifies that a permanent voice connection be torn down and restarted after the trunk has been OOS for a specified time. |
signal timing oos slave-standby | Specifies that a slave port return to its initial standby state after the trunk has been OOS for a specified time |
signal timing oos suppress-all | Configures the router or concentrator to stop sending voice and signaling packets to the network if it detects a transmit OOS signaling pattern from the PBX for a specified time. |
signal timing oos suppress-voice | Configures the router or concentrator to stop sending voice packets to the network if it detects a transmit OOS signaling pattern from the PBX for a specified time. |
signal timing oos timeout | Changes the delay time between the loss of signaling packets from the network and the start time for the OOS state. |
Specifies the length of time before voice traffic is stopped after a trunk goes into the idle state. | |
signal-type | Sets the signaling type to be used when connecting to a dial peer. |
voice class permanent | Creates a voice class for a Cisco trunk or FRF.11 trunk. |
voice-class permanent | Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer. |
To specify which signaling pattern is sent to the PBX when the far-end keepalive message is lost or AIS is received from the far end, use the signal sequence oos voice-class configuration command. Use the no form of this command to restore the default value.
signal sequence oos {no-action | idle-only | oos-only | both}
no-action | No signaling pattern is sent. |
idle-only | Only the idle signaling pattern is sent. |
oos-only | Only the out-of-service (OOS) signaling pattern is sent. |
both | Both idle and OOS signaling patterns are sent. This is the default value. |
Both idle and OOS signal patterns are sent.
Voice-class configuration
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced on the Cisco MC3810. |
Before configuring the idle or OOS signal patterns to be sent, 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. After you finish defining the voice class, you will assign it to a dial peer.
Use this command to specify which signaling pattern(s) to send. Use the signal pattern idle receive or the signal pattern oos receive command to define the bit patterns of the signaling patterns if other than the defaults.
The following example defines voice class 10, sets the signal sequence oos to send only the idle signal pattern to the PBX, and applies the voice class configuration to VoFR dial peer 100.
router(config)#voice class permanent 10 router(config-class)# signal keepalive 3
router(config-class)# signal sequence oos idle-only
router(config-class)# signal timing idle suppress-voice 5 router(config-class)# exit router(config)# dial-peer voice 100 vofr router(config-dial-peer)# voice-class permanent 10 router(config-dial-peer)# signal-type transparent
| Command | Description |
dial-peer voice | Enters dial-peer configuration mode and specifies a dial peer type. |
signal keepalive | Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks. |
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks. | |
Specifies the length of time before the router stops sending voice packets after a trunk goes into the idle state. | |
signal timing oos restart | Specifies that a permanent voice connection be torn down and restarted after the trunk has been OOS for a specified time. |
signal timing oos slave-standby | Specifies that a slave port return to its initial standby state after the trunk has been OOS for a specified time |
signal timing oos suppress-all | Configures the router or concentrator to stop sending voice and signaling packets to the network if it detects a OOS signaling pattern from the PBX for a specified time. |
signal timing oos suppress-voice | Configures the router or concentrator to stop sending voice packets to the network if it detects a transmit OOS signaling pattern from the PBX for a specified time. |
signal timing oos timeout | Changes the delay time between the loss of signaling packets from the network and the start time for the OOS state. |
signal-type | Sets the signaling type to be used when connecting to a dial peer. |
voice class permanent | Creates a voice class for a Cisco trunk or FRF.11 trunk. |
voice-class permanent | Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer. |
To specify the length of time before the router stops sending voice packets after a trunk goes into the idle state (no call in progress), use the signal timing idle suppress-voice voice-class configuration command. Use the no form of this command to restore the default value.
signal timing idle suppress-voice seconds
seconds | Duration of the idle state in seconds before the transmission of voice packets is stopped. The range is 0 to 65535. |
The router or concentrator continues to send voice packets when the trunk is idle.
Voice-class configuration
| Release | Modification |
|---|---|
12.0(3)XG | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was modified to simplify the configuration process. |
Before configuring the signal timing idle suppress-voice timer, 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. After you finish defining the voice class, you will assign it to a dial peer.
This command is used when the signal-type command is set to transparent in the dial peer for the Cisco trunk or FRF.11 trunk connection. When the router or concentrator stops sending voice packets after the specified time, signaling packets continue to be sent.
To detect an idle trunk state, the router or concentrator monitors both transmit and receive signaling for the idle transmit and idle receive signaling patterns. These can be configured by the signal pattern idle transmit or signal pattern idle receive command, or they can be the defaults. The default idle receive pattern is the idle pattern of the local voice port. The default idle transmit pattern is the idle pattern of the far-end voice port.
The following example defines voice class 10, sets the idle detection time to 5 seconds, configures the trunk to use the default transmit and receive idle signal patterns, and applies the voice class configuration to VoFR dial peer 100.
router(config)#voice class permanent 10 router(config-class)# signal keepalive 3
router(config-class)# signal timing idle suppress-voice 5 router(config-class)# exit router(config)# dial-peer voice 100 vofr router(config-dial-peer)# voice-class permanent 10 router(config-dial-peer)# signal-type transparent
| Command | Description |
dial-peer voice | Enters dial-peer configuration mode and specifies a dial peer type. |
signal keepalive | Configures the keepalive signaling packet interval for Cisco trunks and FRF.11 trunks. |
Configures the ABCD bit pattern for Cisco trunks and FRF.11 trunks. | |
Specifies which signaling pattern is sent to the PBX when the far-end keepalive message is lost or AIS is received from the far end. | |
Specifies the length of time before the router stops sending voice packets after a trunk goes into the idle state. | |
signal timing oos restart | Specifies that a permanent voice connection be torn down and restarted after the trunk has been OOS for a specified time. |
signal timing oos slave-standby | Specifies that a slave port return to its initial standby state after the trunk has been OOS for a specified time |
signal timing oos suppress-all | Configures the router or concentrator to stop sending voice and signaling packets to the network if it detects a transmit OOS signaling pattern from the PBX for a specified time. |
signal timing oos timeout | Changes the delay time between the loss of signaling packets from the network and the start time for the OOS state. |
signal-type | Sets the signaling type to be used when connecting to a dial peer. |
voice class permanent | Creates a voice class for a Cisco trunk or FRF.11 trunk. |
voice-class permanent | Assigns a previously-configured voice class for a Cisco trunk or FRF.11 trunk to a dial peer. |
To set the signaling type to be used when connecting to a dial peer, use the signal-type command from dial-peer configuration mode. To return to the default signal-type, use the no form of this command.
signal-type {cas | cept | ext-signal | transparent}
cas | North American EIA-464 Channel-Associated Signaling (robbed bit signaling). If the Digital T1 Packet Voice Trunk Network Module is installed, this option may not be available. |
cept | Provides a basic E1 ABCD signaling protocol. Used primarily for E&M interfaces. When used with FXS/FXO interfaces, this protocol is equivalent to MELCAS. |
ext-signal | External signaling. The DSP does not generate any signaling frames. Use this option when there is an external signaling channel, for example, CCS, or when you need to have a permanent "dumb" voice pipe. |
transparent | On the Cisco MC3810, selecting this option produces different results depending on whether you are using a digital voice module (DVM) or an analog voice module (AVM). For a DVM: The ABCD signaling bits are copied from or transported through the T1/E1 interface "transparently" without modification or interpretation. This enables the MC3810 to handle arbitrary or unknown signaling protocols. For an AVM: It is not possible to provide "transparent" behavior because the Cisco MC3810 must interpret the signaling information in order to read and write the correct state to the analog hardware. This option is mapped to be equal to "cas." |
cas
Dial-peer configuration
| Release | Modification |
|---|---|
12.0(3)XG | This command was introduced. |
12.0(4)T | Support was added for the Cisco 7200 series routers. |
12.0(7)XK | In previous releases, the cept and transparent options were only supported on the Cisco MC3810. Beginning in this release, these options are supported on the Cisco 2600, Cisco 3600 and Cisco 7200 routers. |
This command applies to VoFR and VoATM dial peers. It is used with permanent connections only (Cisco trunks and FRF.11 trunks), not with switched calls.
This command is used to inform the local telephony interface of the type of signaling it should expect to receive from the far-end dial peer. To turn signaling off at this dial peer, select the ext-signal option. If signaling is turned off and there are no external signaling channels, a "hot" line exists, enabling this dial peer to connect to anything at the far end.
When you connect an FXS to another FXS, or if you have anything other than an FXS/FXO or E&M/E&M pair, the appropriate signaling type on Cisco 2600 series and 3600 series routers is ext-signal (disabled).
If you have a digital E1 connection at the remote end that is running cept/MELCAS signaling and you then trunk that across to an analog port, you should make sure that you configure both ends for the cept signal-type.
If you have a T1 or E1 connection at both ends and the T1/E1 is running a signaling protocol that is neither EIA-464 or cept/MELCAS, you may want to configure the signal-type for the transparent option in order to pass through the signaling.
The following example shows how to disable signaling on a Cisco 2600 series or 3600 series router or on an MC3810 concentrator for VoFR dial peer 200, starting from global configuration mode:
router(config)# dial-peer voice 200 vofr
router(config-dial-peer)# signal-type ext-signal
router(config-dial-peer)#
| Command | Description |
codec (dial-peer) | Specifies the voice coder rate of speech for a dial peer. |
connection | Specifies the connection mode for a voice port. |
destination-pattern | Specifies the telephone number associated with a dial peer. |
dtmf-relay | Enables the DSP to generate FRF.11 Annex A frames for a dial peer. |
preference | Enables the preferred dial peer to be selected when multiple dial peers within a hunt group are matched for a dial string. |
session protocol | Establishes the VoFR protocol for calls between local and remote routers. |
session target | Specifies a network-specific address for a dial peer. |
sequence-numbers | Enables the generation of sequence numbers in each frame generated by the DSP. |
To test detector-related functions on a voice port, use the test voice port detector privileged EXEC command.
For the Cisco 2600 and 3600 series with analog voice ports:
test voice port slot/subunit/port detector {m-lead | battery-reversal | ring | tip-ground | ring-ground | ring-trip} {on | off | disable}For the Cisco 2600 and 3600 series with digital voice ports:
test voice port slot/port:ds0-group detector {m-lead | battery-reversal | ring | tip-ground | ring-ground | ring-trip} {on | off | disable}For the Cisco MC3810 series with analog voice ports:
test voice port slot/port detector {m-lead | battery-reversal | ring | tip-ground | ring-ground | ring-trip} {on | off | disable}For the Cisco MC3810 series with digital voice ports:
test voice port slot:ds0-group detector {m-lead | battery-reversal | ring | tip-ground | ring-ground | ring-trip} {on | off | disable}For the Cisco 2600 and 3600 series with analog voice ports:
slot/subunit/port | Tests the voice port you specify with the slot/subunit/port designation. slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform. subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. port specifies an analog voice port number. Valid entries are 0 and 1. |
For the Cisco 2600 and 3600 series with digital voice ports:
slot/port:ds0-group | Tests the voice port you specify with the slot/port:ds0-group designation. slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform. port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For the Cisco MC3810 series with analog voice ports:
slot/port | Tests the voice port you specify with the slot/port designation. slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810. port specifies an analog voice port number. Valid entries are 1 to 6. |
For the Cisco MC3810 series with digital voice ports:
slot:ds0-group | Tests the voice port you specify with the slot:ds0-group designation. slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1). ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For all platforms:
m-lead | Forces the E&M m-lead detector to the specified state |
loop | Forces the FXO loop detector to the specified state |
battery-reversal | Forces the FXO battery-reversal detector to the specified state |
ring | Forces the FXO ringing detector to the specified state |
tip-ground | Forces the FXO tip-ground detector to the specified state |
ring-ground | Forces the FXS ring-ground detector to the specified state |
ring-trip | Forces the FXS ring-trip detector to the specified state |
on | Forces the selected item to the on state |
off | Forces the selected item to the off state |
disable | Ends the forced state for the selected item |
Privileged EXEC
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced. |
Use the test voice port detector privileged EXEC command to force a detector into specific states for testing. For each signaling type (E&M, FXO, FXS), only the applicable keywords are displayed. When you are finished testing, be sure to enter the disable command to end the forced state. The disable keyword is available only if a test condition is already activated.
The following example forces the tip-ground detector to the off state on an FXO voice port (1/3) on a Cisco MC3810, and ends any call in progress:
router# test voice port 1/3 detector tip-ground off
The following example ends the forced off state on an FXO voice port (1/3) on a Cisco MC3810:
router# test voice port 1/3 detector tip-ground disable
The following example forces the ring-trip detector to the on state on an FXS port (0/0/1) on a Cisco 3600 series router, and should start a call:
router# test voice port 0/0/1 detector ring-trip on
The following example ends the forced on state on an FXS port (0/0/1) on a Cisco 3600 series router:
router# test voice port 0/0/1 detector ring-trip disable
| Command | Description |
Performs loopback testing on a voice port. | |
Injects a test tone into a voice port. | |
Tests relay-related functions on a voice port. | |
Forces a voice port into fax or voice mode. |
To inject a test tone into a voice port, use the test voice port inject-tone privileged EXEC command.
For the Cisco 2600 and 3600 series with analog voice ports:
test voice port slot/subunit/port inject-tone {local | network} {1000hz | 2000hz | 200hz | 3000hz | 300hz | 3200hz | 3400hz | 500hz | quiet | disable}For the Cisco 2600 and 3600 series with digital voice ports:
test voice port slot/port:ds0-group inject-tone {local | network} {1000hz | 2000hz | 200hz | 3000hz | 300hz | 3200hz | 3400hz | 500hz | quiet | disable}For the Cisco MC3810 series with analog voice ports:
test voice port slot/port inject-tone {local | network} {1000hz | 2000hz | 200hz | 3000hz | 300hz | 3200hz | 3400hz | 500hz | quiet | disable}For the Cisco MC3810 series with digital voice ports:
test voice port slot:ds0-group inject-tone {local | network} {1000hz | 2000hz | 200hz | 3000hz | 300hz | 3200hz | 3400hz | 500hz | quiet | disable}For the Cisco 2600 and 3600 series with analog voice ports:
slot/subunit/port | Tests the voice port you specify with the slot/subunit/port designation. slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform. subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. port specifies an analog voice port number. Valid entries are 0 and 1. |
For the Cisco 2600 and 3600 series with digital voice ports:
slot/port:ds0-group | Tests the voice port you specify with the slot/port:ds0-group designation. slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform. port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For the Cisco MC3810 series with analog voice ports:
slot/port | Tests the voice port you specify with the slot/port designation. slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810. port specifies an analog voice port number. Valid entries are 1 to 6. |
For the Cisco MC3810 series with digital voice ports:
slot:ds0-group | Tests the voice port you specify with the slot:ds0-group designation. slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1). ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For all platforms:
local | Directs the injected tone toward the local interface (near end) |
network | Directs the injected tone toward the network (far end) |
1000hz | Injects a 1-kilohertz test tone |
2000hz | Injects a 2-kilohertz test tone |
200hz | Injects a 200-hertz test tone |
3000hz | Injects a 3-kilohertz test tone |
300hz | Injects a 300-hertz test tone |
3200hz | Injects a 3.2-kilohertz test tone |
3400hz | Injects a 3.4-kilohertz test tone |
500hz | Injects a 500-hertz test tone |
quiet | Injects a quiet tone |
disable | Ends test tone |
Privileged EXEC
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced. |
Use the test voice port inject-tone privileged EXEC and to inject a test tone or to end a test tone. A call must be established on the voice port under test. When you are finished testing, be sure to enter the disable command to end the test tone. The disable keyword is available only if a test condition is already activated.
When you enter the disable command, you must enter a direction (either network or local); however, you can enter either direction, regardless of which direction you entered to inject the test tone.
The following example injects a 1-kilohertz test tone into voice port 1/1, directed toward the network (far end), on a Cisco MC3810:
router# test voice port 1/1 inject-tone network 1khz
The following example removes the test tone from port 0/0/1 on a Cisco 3600 series router:
router# test voice port 0/0/1 inject-tone network disable
or
router# test voice port 0/0/1 inject-tone local disable
| Command | Description |
Tests detector-related functions on a voice port | |
Performs loopback testing on a voice port. | |
Tests relay-related functions on a voice port. | |
Forces a voice port into fax or voice mode. |
To perform loopback testing on a voice port, use the test voice port loopback privileged EXEC command.
For the Cisco 2600 and 3600 series with analog voice ports:
test voice port slot/subunit/port loopback {local | network | disable}For the Cisco 2600 and 3600 series with digital voice ports:
test voice port slot/port:ds0-group loopback {local | network | disable}For the Cisco MC3810 series with analog voice ports:
test voice port slot/port loopback {local | network | disable}For the Cisco MC3810 series with digital voice ports:
test voice port slot:ds0-group loopback {local | network | disable}For the Cisco 2600 and 3600 series with analog voice ports:
slot/subunit/port | Tests the voice port you specify with the slot/subunit/port designation. slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform. subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. port specifies an analog voice port number. Valid entries are 0 and 1. |
For the Cisco 2600 and 3600 series with digital voice ports:
slot/port:ds0-group | Tests the voice port you specify with the slot/port:ds0-group designation. slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform. port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For the Cisco MC3810 series with analog voice ports:
slot/port | Tests the voice port you specify with the slot/port designation. slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810. port specifies an analog voice port number. Valid entries are 1 to 6. |
For the Cisco MC3810 series with digital voice ports:
slot:ds0-group | Tests the voice port you specify with the slot:ds0-group designation. slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1). ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For all platforms:
local | Forces a loopback at the voice port toward the customer premises equipment (CPE) |
network | Forces a loopback at the voice port toward network |
disable | Ends forced loopback |
Privileged EXEC
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced. |
Use the test voice port loopback privileged EXEC command to initiate or end a loopback at a voice port. A call must be established on the voice port under test. When you are finished testing, be sure to enter the disable command to end the forced loopback. The disable keyword is available only if a test condition is already activated.
The following example forces a loopback toward the CPE on voice port 1/1 on a Cisco MC3810:
router# test voice port 1/1 loopback local
The following example ends a forced loopback on port 0/0/1 on a Cisco 3600 series router:
router# test voice port 0/0/1 loopback disable
| Command | Description |
Tests detector-related functions on a voice port. | |
Injects a test tone into a voice port. | |
Tests relay-related functions on a voice port. | |
Forces a voice port into fax or voice mode. |
To test relay-related functions on a voice port, use the test voice port relay privileged EXEC command.
For the Cisco 2600 and 3600 series with analog voice ports:
test voice port slot/subunit/port relay {e-lead | loop | ring-ground | battery-reversal | power-denial | ring | tip-ground} {on | off | disable}For the Cisco 2600 and 3600 series with digital voice ports:
test voice port slot/port:ds0-group relay {e-lead | loop | ring-ground | battery-reversal | power-denial | ring | tip-ground} {on | off | disable}For the Cisco MC3810 series with analog voice ports:
test voice port slot/port relay {e-lead | loop | ring-ground | battery-reversal | power-denial | ring | tip-ground} {on | off | disable}For the Cisco MC3810 series with digital voice ports:
test voice port slot:ds0-group relay {e-lead | loop | ring-ground | battery-reversal | power-denial | ring | tip-ground} {on | off | disable}For the Cisco 2600 and 3600 series with analog voice ports:
slot/subunit/port | Tests the voice port you specify with the slot/subunit/port designation. slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform. subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. port specifies an analog voice port number. Valid entries are 0 and 1. |
For the Cisco 2600 and 3600 series with digital voice ports:
slot/port:ds0-group | Tests the voice port you specify with the slot/port:ds0-group designation. slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform. port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For the Cisco MC3810 series with analog voice ports:
slot/port | Tests the voice port you specify with the slot/port designation. slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810. port specifies an analog voice port number. Valid entries are 1 to 6. |
For the Cisco MC3810 series with digital voice ports:
slot:ds0-group | Tests the voice port you specify with the slot:ds0-group designation. slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1). ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For all platforms:
e-lead | Forces the E&M e-lead relay to the specified state |
loop | Forces the FXO loop relay to the specified state |
ring-ground | Forces the FXO ring-ground relay to the specified state |
battery-reversal | Forces the FXO battery-reversal relay to the specified state |
power-denial | Forces the FXS power-denial relay to the specified state |
ring | Forces the FXS ringing relay to the specified state |
tip-ground | Forces the FXS tip-ground relay to the specified state |
on | Forces the selected item to the on state |
off | Forces the selected item to the off state |
disable | Ends the forced state for the selected item |
Privileged EXEC
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced. |
Use the test voice port relay privileged EXEC command to force a relay into specific states for testing. For each signaling type (E&M, FXO, FXS), only the applicable keywords are displayed. When you are finished testing, be sure to enter the disable command to end the forced state. The disable keyword is available only if a test condition is already activated.
The following example forces the E&M e-lead relay to the on state on port 0/0/1 on a Cisco 3600 series router:
router# test voice port 0/0/1 relay e-lead on
The following example ends a forced actuation of the battery-reversal relay on an FXS port (0/0/1) on a Cisco 3600 series router:
router# test voice port 0/0/1 relay battery-reversal disable
| Command | Description |
Tests detector-related functions on a voice port | |
Injects a test tone into a voice port. | |
Performs loopback testing on a voice port. | |
Forces a voice port into fax or voice mode. |
To force a voice port into fax mode, use the test voice port switch privileged EXEC command.
For the Cisco 2600 and 3600 series with analog voice ports:
test voice port slot/subunit/port switch {fax | disable}For the Cisco 2600 and 3600 series with digital voice ports:
test voice port slot/port:ds0-group switch {fax | disable}For the Cisco MC3810 series with analog voice ports:
test voice port slot/port switch {fax | disable}For the Cisco MC3810 series with digital voice ports:
test voice port slot:ds0-group switch {fax | disable}For the Cisco 2600 and 3600 series with analog voice ports:
slot/subunit/port | Tests the voice port you specify with the slot/subunit/port designation. slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform. subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. port specifies an analog voice port number. Valid entries are 0 and 1. |
For the Cisco 2600 and 3600 series with digital voice ports:
slot/port:ds0-group | Tests the voice port you specify with the slot/port:ds0-group designation. slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform. port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For the Cisco MC3810 series with analog voice ports:
slot/port | Tests the voice port you specify with the slot/port designation. slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810. port specifies an analog voice port number. Valid entries are 1 to 6. |
For the Cisco MC3810 series with digital voice ports:
slot:ds0-group | Tests the voice port you specify with the slot:ds0-group designation. slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1). ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For all platforms:
fax | Forces a switch to fax mode |
disable | Ends fax mode; switches back to voice mode |
Privileged EXEC
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced. |
Use the test voice port switch privileged EXEC command to force a voice port into fax mode for testing. If no fax data is detected by the voice port, the voice port remains in fax mode for 30 seconds and then reverts automatically to voice mode. After you enter the test voice port switch fax command, you can use the show voice call or show voice call summary command to check whether the voice port is able to operate in fax mode.
The disable command ends the forced mode switch; however, the fax mode ends automatically after 30 seconds. The disable keyword is available only while the voice port is in fax mode.
The following example forces voice port 1/3 on a Cisco MC3810 into fax mode:
router# test voice port 1/3 switch fax
The following example returns voice port 0/0/1 on a Cisco 3600 series router to voice mode:
router# test voice port 0/0/1 switch disable
| Command | Description |
show voice call | Shows the call processing and protocol state-machine information for a voice port. |
show voice call summary | Shows a summary of the call processing and protocol state-machine information for a voice port. |
To configure the timeout value for ringing, use the timeouts ringing voice-port configuration command. Use the no form of this command to restore the default value.
timeouts ringing {seconds | infinity}
seconds | The duration in seconds that a voice port allows ringing to continue if a call is not answered. The range is 5 to 60000. |
infinity | Ringing continues until the caller goes on hook. |
180 seconds
Voice-port configuration
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced. |
This command provides the capability to limit the length of time that a caller can continue ringing a telephone when there is no answer.
The following example configures voice port 1/1 on a Cisco MC3810 to allow ringing for 600 seconds:
router(config)# voice-port 1/1 router(config-voiceport)# timeouts ringing 600
The following example configures voice port 0/0/1 on a Cisco 3600 to allow ringing for 600 seconds:
router(config)# voice-port 0/0/1 router(config-voiceport)# timeouts ringing 600
| Command | Description |
timeouts initial | Configures the initial-digit timeout value for a voice port. |
timeouts interdigit | Configures the interdigit timeout value for a voice port. |
To configure the delay timeout before the system starts the process for releasing voice ports, use the timeouts wait-release voice-port configuration command. Use the no form of this command to restore the default value.
timeouts wait-release {seconds | infinity}
seconds | The duration in seconds that a voice port stays in the call-failure state while the Cisco router or concentrator sends a busy tone, reorder tone, or an out-of-service tone to the port. The range is 3 to 3600. |
infinity | The voice port is never released as long as the call-failure state remains. |
30 seconds
Voice-port configuration
| Release | Modification |
|---|---|
11.3(1) MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
Use this command to limit the time a voice port can be held in a call failure state. After the timeout, the release sequence is enabled.
You can also use this command for voice ports with FXS loop-start signaling, to specify the time allowed for a caller to hang up before the voice port goes into the parked state.
The following example configures voice port 1/1 on a Cisco MC3810 to stay in the call-failure state for 180 seconds while a busy tone, reorder tone, or out-of-service tone is sent to the voice port:
router(config)# voice-port 1/1 router(config-voiceport)# timeouts wait-release 180
The following example configures voice port 0/0/1 on a Cisco 3600 to stay in the call-failure state for 180 seconds while a busy tone, reorder tone, or out-of-service tone is sent to the voice port:
router(config)# voice-port 0/0/1 router(config-voiceport)# timeouts wait-release 180
| Command | Description |
timeouts initial | Configures the initial-digit timeout value for a voice port. |
timeouts interdigit | Configures the interdigit timeout value for a voice port. |
To specify the guard-out duration of an FXO voice port, use the timing guard-out voice-port configuration command. Use the no form of this command to restore the default value.
timing guard-out milliseconds
milliseconds | Duration in milliseconds of the guard-out period. The range is 300 to 3000. The default is 2000. |
2000 milliseconds
Voice-port configuration
| Release | Modification |
|---|---|
11.3(1)MA5 | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
This command applies to the Cisco 2600, 3600, and MC3810 platforms.
This command is supported on FXO voice ports only.
The following example configures the timing guard-out duration on a Cisco MC3810 voice port to 1000 milliseconds:
router(config)# voice-port 1/1 router(config-voiceport)# timing guard-out 1000
The following example configures the timing guard-out duration on a Cisco 2600 or 3600 voice port to 1000 milliseconds:
router(config)# voice-port 1/0/0 router(config-voiceport)# timing guard-out 1000
To specify the percentage of the break period for dialing pulses for a voice port, use the timing percentbreak voice-port configuration command. Use the no form of this command to reset the default value.
timing percentbreak percent
percent | Percentage of the break period for dialing pulses. Valid entries are numbers 20 to 80. The default is 50. |
50 percent
Voice-port configuration
| Release | Modification |
|---|---|
11.3(1) MA4 | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 2600 and 3600 series routers. |
This command is supported on FXO and E&M voice ports only.
The following example configures the break period percentage on a Cisco MC3810 voice port to 30 percent:
router(config)# voice-port 1/1 router(config-voiceport)# timing percentbreak 30
The following example configures the break period percentage on a Cisco 2600 or 3600 voice port to 30 percent:
router(config)# voice-port 0/0/1 router(config-voiceport)# timing percentbreak 30
| Command | Description |
timing pulse | Configures the pulse dialing rate for a voice port. |
timing pulse-interdigit | Configures the pulse inter-digit timing for a voice port. |
peak-rate | The peak information rate (PIR) of the voice connection in kbps. The range is 56 to 10000. |
average-rate | The average information rate (AIR) of the voice connection in kbps. The range is 1 to 56. |
burst | Burst size in number of cells. The range is 0 to 65536. |
No vbr-rt settings are configured.
ATM virtual circuit configuration
| Release | Modification |
|---|---|
12.0 | This command was introduced on the Cisco MC3810. |
12.0(7)XK | Support for this command was extended to the Cisco 3600 series. |
The vbr-rt command configures traffic shaping between voice and data PVCs. Traffic shaping is required so that the carrier does not discard calls. To configure voice and data traffic shaping, you must configure the peak, average, and burst options for voice traffic. Configure the burst value if the PVC will be carrying bursty traffic. The peak, average, and burst values are needed so the PVC can effectively handle the bandwidth for the number of voice calls. To calculate the minimum peak, average, and burst values for the number of voice calls, use the following calculations:
The following example configures the traffic shaping rate for ATM PVC 20 on a Cisco 3600. In the example, the peak, average and burst rates are calculated based on a maximum of 20 calls on the PVC.
router(config-if)# pvc 20 router(config-if-atm-pvc)# encapsulation aal5mux voice router(config-if-atm-pvc)# vbr-rt 640 320 80
| Command | Description |
Configures the ATM adaptation layer (AAL) and encapsulation type for an ATM PVC class |
To enable Voice over Frame Relay (VoFR) on a specific DLCI and to configure specific subchannels on that DLCI, use the vofr command from Frame Relay DLCI configuration mode. Use the no form of the command to disable VoFR on a specific DLCI.
For switched calls:
vofr [data cid] [call-control [cid]]For switched calls to Cisco MC3810 concentrators running Cisco IOS releases before 12.0(7)XK:
vofr [cisco]For Cisco-trunk permanent calls:
vofr data cid call-control cidFor Cisco-trunk permanent calls to Cisco MC3810 concentrators running Cisco IOS releases before 12.0(7)XK:
vofr ciscoFor FRF-11 trunk calls:
vofr [data cid] [call-control cid]
data | (Required for Cisco-trunk permanent calls. Optional for switched calls.) Used to select a subchannel (CID) for data other than the default subchannel, which is 4. |
cid | (Optional) Specifies the subchannel to be used for data. Valid values are from 4 to 255; the default is 4. If data is specified, enter a valid CID. |
call-control | (Optional) Used to specify that a subchannel will be reserved for call-control signaling. This option is not supported on the Cisco MC3810. |
cid | (Optional) Specifies the subchannel to be used for call-control signaling. Valid values are from 4 to 255; the default is 5. If you specify call-control and you do not enter a CID, the default CID is used. |
cisco | (Optional) Cisco proprietary voice encapsulation for VoFR with data is carried on CID 4 and call-control on CID 5. This option is required when configuring switched calls or Cisco trunks to Cisco MC3810 concentrators running Cisco IOS releases before 12.0(7)XK. If configuring switched calls or Cisco trunks to Cisco MC3810 concentrators running Cisco IOS release 12.0(7)XK and later releases, do not use this option. |
Disabled
Frame Relay DLCI
| Release | Modification |
|---|---|
12.0(3)XG and 12.0(4)T | This command was introduced. |
12.0(7)XK | The use of the cisco option was modified. Beginning in this release, use the cisco option only when configuring connections to Cisco MC3810 concentrators running Cisco IOS releases before 12.0(7)XK. |
Table 10 lists the different options of the vofr command and which combination of options is used.
| Type of Call | vofr Command Combination to Use |
|---|---|
Switched call | vofr [data cid] |
Switched call | vofr cisco2
|
Cisco-trunk | vofr data cid |
Cisco-trunk | vofr cisco |
FRF.11 trunk call (private-line) to other routers supporting VoFR | vofr [data cid] [call-control cid]3 |
If you select the "data" option, enter a numeric value to complete the command. If you select the "call-control" option, you do not enter a numeric value if you wish to accept the default call-control subchannel. See the following examples for clarification.
Usage Restrictions for Cisco IOS Releases Prior to 12.0(7)XK
This section describes restrictions for using the vofr command in releases prior to Cisco IOS Release 12.0(7)XK. Beginning in Cisco IOS Release 12.0(7)XK, these restrictions no longer apply.
When you use the vofr command without the cisco option, all subchannels on the DLCI are configured for FRF.11 encapsulation. If you enter the vofr command is entered without any keywords or arguments, the data subchannel is CID 4 and there is no call-control subchannel.
Table 11 describes special conditions and restrictions for the use of the vofr command on the Cisco MC3810.
| Type of Call | Conditions and Restrictions |
|---|---|
FRF.11 trunks | 1. Do NOT use cisco option or call-control option. 2. Use vofr or vofr data cid. |
Cisco trunks | 1. Must use vofr cisco. |
switched-vofr | 1. Must use vofr cisco. |
If you select the "data" option, enter a numeric value to complete the command. If you select the "call-control" option, you do not enter a numeric value if you wish to accept the default call-control subchannel. See the following examples for clarification.
When you use the vofr command on a Cisco MC3810 without the "cisco" option, switched calls are not permitted. You can only make permanent FRF.11-trunk calls.
The following example shows how to enable VoFR on Serial 1/1, DLCI 100 on a Cisco 2600 series, 3600 series, or 7200 series router or on an MC3810 concentrator, starting from global configuration mode:
router(config)# interface serial 1/1
router(config-if)# frame-relay interface-dlci 100
router(config-fr-dlci)# vofr
router(config-fr-dlci)#
The above example configures CID 4 for data; no call-control CID is defined.
To configure CID 4 for data and CID 5 for call-control (both defaults), enter the following command:
router(config-fr-dlci)# vofr call-control router(config-fr-dlci)#
To configure CID10 for data and CID 15 for call-control, enter the following command:
router(config-fr-dlci)# vofr data 10 call-control 15 router(config-fr-dlci)#
To configure CID 4 for data and CID 15 for call-control, enter the following command:
router(config-fr-dlci)# vofr call-control 15 router(config-fr-dlci)#
To configure CID 10 for data and CID 5 for call-control, enter the following command:
router(config-fr-dlci)# vofr data 10 call-control router(config-fr-dlci)#
To configure CID 10 for data with no call-control, enter the following command:
router(config-fr-dlci)# vofr data 10 router(config-fr-dlci)#
To configure a Cisco router or MC3810 for a VoFR application with an older release of the MC3810 (before Release 12.0(3)XG), enter the following command:
router(config-fr-dlci)# vofr cisco router(config-fr-dlci)#
| Command | Description |
frame-relay interface-dlci | Assigns a data link connection identifier (DLCI) to a specified Frame Relay subinterface. |
class | Assigns a VC class to a PVC. |
To configure a voice card and enter voice-card configuration mode, enter the voice-card command.
voice-card slot
slot | On the Cisco 2600 and 3600 platforms: · A value from 0 to 3 that identifies the physical slot in the chassis where the voice card is located. On Cisco MC3810 concentrators with one or two HCMs installed: · Enter 0 only; this applies to the entire chassis. |
Global configuration
| Release | Modification |
|---|---|
12.0(5)XK and 12.0(7)T | The command was introduced for the Cisco 2600 and 3600 series. |
12.0(7)XK | This command was first supported on the Cisco MC3810 series. |
You can configure codec complexity only in voice-card configuration mode. On the Cisco 2600 and 3600 platforms, the slot corresponds to the physical slot in the chassis. On the Cisco MC3810, the slot is always 0, and all changes made in voice-card mode apply to the entire Cisco MC3810. On Cisco MC3810 series concentrators, this command is available only if the chassis is equipped with one or two HCMs.
The following example enters voice-card configuration mode for the voice card in slot 1 on a Cisco 2600 or 3600 router:
router(config)#voice-card 1
router(config-voicecard)#
The following example enters voice-card configuration mode on a Cisco MC3810 concentrator:
router(config)#voice-card 0
router(config-voicecard)#
| Command | Description |
Matches the DSP complexity packaging to the codec(s) to be supported. Codec complexity changes are made in the voice-card configuration mode. |
This command was added in Cisco IOS Release 11.3(1)MA on the Cisco MC3810 for Voice over HDLC. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
This command was added in Cisco IOS Release 11.3(1)MA on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported.
To configure local calls to bypass the digital signal processor (DSP), use the voice local-bypass global configuration command. Use the no form of this command to direct local calls through the DSP.
voice local-bypassThis command has no arguments or keywords.
Local calls bypass the DSP.
Global configuration
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced. |
Local calls (calls between voice ports on a router or concentrator) normally bypass the DSP to minimize use of system resources. Use the no form of this command if you need to direct local calls through the DSP. Input gain and output attenuation can be configured only if calls are directed through the DSP.
The following example configures a Cisco MC3810, 2600, or 3600 to pass local calls through the DSP:
router(config)# no voice local-bypass
| Command | Description |
input gain | Configures receive gain value for a voice port. |
output attenuation | Configures transmit attenuation value for a voice port. |
To change the minimum silence detection time for voice activity detection (VAD), use the voice vad-time global configuration command. Use the no form of this command to restore the default value.
voice vad-time milliseconds
milliseconds | The waiting period in milliseconds before silence detection and suppression of voice-packet transmission. The range is 250 to 65536. The default is 250. |
250 milliseconds
Global configuration
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced on the Cisco 2600, 3600, and MC3810. |
This command affects all voice ports on a router or concentrator, but it does not affect calls already in progress.
You can use this command in transparent CCS applications in which you want VAD to activate when the voice channel is idle, but not during active calls. With a longer silence detection delay, VAD reacts to the silence of an idle voice channel, but not to pauses in conversation.
This command does not affect voice codecs that have ITU-standardized built-in VAD features—for example, G.729B, G.729AB, G.723.1A. The VAD behavior and parameters of these codecs are defined exclusively by the applicable ITU standard.
The following example configures a 20-second delay before VAD silence detection is enabled:
router(config)# voice vad-time 20000
| Command | Description |
vad (dial peer) | Enables voice activity detection on a network dial peer. |
This section documents new, modified and removed commands. All other commands used on these platforms are documented in the Cisco IOS Release 12.0 command reference publications.
The following debug commands have been removed in Cisco IOS Release 12.0(7)XK:
To display the ccfrf11 function calls during call setup and teardown, use the debug ccfrf11 session command from privileged EXEC mode. Use the no form of this command to turn off the debug function.
debug ccfrf11 sessionThis command has no keywords or arguments.
Privileged EXEC
| Release | Modification |
|---|---|
12.0(3)XG and 12.0(4)T | This command was introduced on the Cisco 2600 and Cisco 3600 series routers. |
12.0(7)XK | This command was first supported on the Cisco MC3810. |
Use this command to display debug information about the various FRF.11 VoFR service provider interface (SPI) functions. Note that this debug command does not display any information regarding the proprietary Cisco switched-VoFR SPI.
This debug is only useful when the session protocol is "frf11-trunk."
The following example shows sample output from the debug ccfr11 session command:
router# debug ccfrf11 session
INCOMING CALL SETUP (port setup for answer-mode):
*Mar 6 18:04:07.693:ccfrf11_process_timers:scb (0x60EB6040) timer (0x60EB6098) expired
*Mar 6 18:04:07.693:Setting accept_incoming to TRUE
*Mar 6 18:04:11.213:ccfrf11_incoming_request:peer tag 800:callingNumber=+2602100,
calledNumber=+3622110
*Mar 6 18:04:11.213:ccfrf11_initialize_ccb:preffered_codec set(-1)(0)
*Mar 6 18:04:11.213:ccfrf11_evhandle_incoming_call_setup_request:calling +2602100,
called +3622110 Incoming Tag 800
*Mar 6 18:04:11.217:ccfrf11_caps_ind:PeerTag = 800
*Mar 6 18:04:11.217: codec(preferred) = 4, fax_rate = 2, vad = 2
*Mar 6 18:04:11.217: cid = 30, config_bitmask = 0, codec_bytes = 20, signal_type=2
*Mar 6 18:04:11.217: required_bandwidth 8192
*Mar 6 18:04:11.217:ccfrf11_caps_ind:Bandwidth reservation of 8192 bytes succeeded.
*Mar 6 18:04:11.221:ccfrf11_evhandle_call_connect:Entered
CALL SETUP (MASTER):
5d22h:ccfrf11_call_setup_request:Entered
5d22h:ccfrf11_evhandle_call_setup_request:Entered
5d22h:ccfrf11_initialize_ccb:preffered_codec set(-1)(0)
5d22h:ccfrf11_evhandle_call_setup_request:preffered_codec set(9)(24)
5d22h:ccfrf11_call_setup_trunk:subchannel linking successful
5d22h:ccfrf11_caps_ind:PeerTag = 810
5d22h: codec(preferred) = 512, fax_rate = 2, vad = 2
5d22h: cid = 30, config_bitmask = 1, codec_bytes = 24, signal_type=2
5d22h: required_bandwidth 6500
5d22h:ccfrf11_caps_ind:Bandwidth reservation of 6500 bytes succeeded.
CALL TEARDOWN:
*Mar 6 18:09:14.805:ccfrf11_call_disconnect:peer tag 0
*Mar 6 18:09:14.805:ccfrf11_evhandle_call_disconnect:Entered
*Mar 6 18:09:14.805:ccfrf11_call_cleanup:freeccb 1, call_disconnected 1
*Mar 6 18:09:14.805:ccfrf11_call_cleanup:Setting accept_incoming to FALSE and starting
incoming timer
*Mar 6 18:09:14.809:timer 2:(0x60EB6098)starts - delay (70000)
*Mar 6 18:09:14.809:ccfrf11_call_cleanup:Alive timer stopped
*Mar 6 18:09:14.809:timer 1:(0x60F64104) stops
*Mar 6 18:09:14.809:ccfrf11_call_cleanup:Generating Call record
*Mar 6 18:09:14.809:cause=10 tcause=10 cause_text="normal call clearing."
*Mar 6 18:09:14.809:ccfrf11_call_cleanup:Releasing 8192 bytes of reserved bandwidth
*Mar 6 18:09:14.809:ccfrf11_call_cleanup:ccb 0x60F6404C, vdbPtr 0x610DB7A4
freeccb_flag=1, call_disconnected_flag=1
| Command | Description |
Displays the ccswvoice function calls during call setup and teardown. | |
Displays the ccswvoice function calls during call setup and teardown. | |
Displays the first 10 bytes (including header) of selected VoFR subframes for the interface. |
To display the ccswvoice function calls during call setup and teardown, use the debug ccswvoice vooatm-debug command from privileged EXEC mode. Use the no form of this command to turn off the debug function.
debug ccswvoice atm-debugThis command has no arguments or keywords.
Privileged EXEC
| Release | Modification |
|---|---|
11.3(1)MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 3600 series. |
This command should be used when attempting to troubleshoot a VoATM call that uses the "cisco-switched" session protocol.This command provides the same information as the debug ccswvoice voatm-session command, but includes additional debugging information relating to the calls.
The following example shows sample output from the debug ccswvoice voatm-debug command:
router# debug ccswvoice voatm-debug 2w2d: ccswvoice: callID 529927 pvcid -1 cid -1 state NULL event O/G SETUP 2w2d: ccswvoice_out_callinit_setup: callID 529927 using pvcid 1 cid 15 2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state O/G INIT event I/C PROC 2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state O/G PROC event I/C ALERTccfrf11_caps_ind: codec(preferred) = 1 2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state O/G ALERT event I/C CONN 2w2d: ccswvoice_bridge_drop: dropping bridge calls src 529927 dst 529926 pvcid 1 cid 15 state ACTIVE 2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state ACTIVE event O/G REL 2w2d: ccswvoice: callID 529927 pvcid 1 cid 15 state RELEASE event I/C RELCOMP 2w2d: ccswvoatm_store_call_history_entry: cause=10 tcause=10 cause_text=normal call clearing.
| Command | Description |
Displays the ccswvoice function calls during call setup and teardown. |
To display the ccswvoice function calls during call setup and teardown, use the debug ccswvoice voatm-session command from privileged EXEC mode. Use the no form of this command to turn off the debug function.
debug ccswvoice voatm-sessionThis command has no arguments or keywords.
Privileged EXEC
| Release | Modification |
|---|---|
11.3(1)MA | This command was introduced on the Cisco MC3810. |
12.0(7)XK | This command was first supported on the Cisco 3600 series. |
Use this command to show the state transitions of the cisco-switched-voatm state machine as a call is processed. This command should be used when attempting to troubleshoot a VoATM call that uses the "cisco-switched" session protocol.
The following example shows sample output from the debug ccswvoice voatm-session command:
router# debug ccswvoice voatm-session 2w2d: ccswvoice: callID 529919 pvcid -1 cid -1 state NULL event O/G SETUP 2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state O/G INIT event I/C PROC 2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state O/G PROC event I/C ALERT 2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state O/G ALERT event I/C CONN 2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state ACTIVE event O/G REL 2w2d: ccswvoice: callID 529919 pvcid 1 cid 11 state RELEASE event I/C RELCOMP
| Command | Description |
Displays the ccswvoice function calls during call setup and teardown. |
To display the ccswvoice function calls during call setup and teardown, use the debug ccswvoice vofr-debug command from privileged EXEC mode. Use the no form of this command to turn off the debug function.
debug ccswvoice vofr-debugThis command has no arguments or keywords.
Privileged EXEC
| Release | Modification |
|---|---|
12.0(3)XG and 12.0(4)T | This command was introduced on the Cisco 2600 and Cisco 3600 series routers. |
12.0(7)XK | This command was first supported on the Cisco MC3810. |
Use this command when troubleshooting a VoFR call that uses the "cisco-switched" session protocol. This command provides the same information as the debug ccswvoice vofr-session command, but includes additional debugging information relating to the calls.
The following example shows sample output from the debug ccswvoice vofr-debug command:
router# debug ccswvoice vofr-debug
CALL TEARDOWN:
3640_vofr(config-voiceport)#
*Mar 1 03:02:08.719:ccswvofr_bridge_drop:dropping bridge calls src 17 dst 16 dlci 100
cid 9 state ACTIVE
*Mar 1 03:02:08.727:ccswvofr:callID 17 dlci 100 cid 9 state ACTIVE event O/G REL
*Mar 1 03:02:08.735:ccswvofr:callID 17 dlci 100 cid 9 state RELEASE event I/C RELCOMP
*Mar 1 03:02:08.735:ccswvofr_store_call_history_entry:cause=22 tcause=22
cause_text=no circuit.
3640_vofr(config-voiceport)#
CALL SETUP (outgoing):
*Mar 1 03:03:22.651:ccswvofr:callID 23 dlci -1 cid -1 state NULL event O/G SETUP
*Mar 1 03:03:22.651:ccswvofr_out_callinit_setup:callID 23 using dlci 100 cid 10
*Mar 1 03:03:22.659:ccswvofr:callID 23 dlci 100 cid 10 state O/G INIT event I/C PROC
*Mar 1 03:03:22.667:ccswvofr:callID 23 dlci 100 cid 10 state O/G PROC event I/C CONN
ccfrf11_caps_ind:codec(preferred) = 0
| Command | Description |
Displays the ccfrf11 function calls during call setup and teardown. | |
Displays the ccswvoice function calls during call setup and teardown. | |
Displays the first 10 bytes (including header) of selected VoFR subframes for the interface. |
To display the ccswvoice function calls during call setup and teardown, use the debug ccswvoice vofr-session command from privileged EXEC mode. Use the no form of this command to turn off the debug function.
debug ccswvoice vofr-sessionThis command has no arguments or keywords.
Privileged EXEC
| Release | Modification |
|---|---|
12.0(3)XG and 12.0(4)T | This command was introduced on the Cisco 2600 and Cisco 3600 series routers. |
12.0(7)XK | This command was first supported on the Cisco MC3810. |
Use this command to show the state transitions of the cisco-switched-vofr state machine as a call is processed, and when attempting to troubleshoot a VoFR call that uses the "cisco-switched" session protocol.
The following example shows sample output from the debug ccswvoice vofr-session command:
router# debug ccswvoice vofr-session CALL TEARDOWN: 3640_vofr(config-voiceport)# *Mar 1 02:58:13.203:ccswvofr:callID 14 dlci 100 cid 8 state ACTIVE event O/G REL *Mar 1 02:58:13.215:ccswvofr:callID 14 dlci 100 cid 8 state RELEASE event I/C RELCOMP 3640_vofr(config-voiceport)# CALL SETUP (outgoing): *Mar 1 02:59:46.551:ccswvofr:callID 17 dlci -1 cid -1 state NULL event O/G SETUP *Mar 1 02:59:46.559:ccswvofr:callID 17 dlci 100 cid 9 state O/G INIT event I/C PROC *Mar 1 02:59:46.567:ccswvofr:callID 17 dlci 100 cid 9 state O/G PROC event I/C CONN 3640_vofr(config-voiceport)#
| Command | Description |
Displays the ccfrf11 function calls during call setup and teardown. | |
Displays the ccswvoice function calls during call setup and teardown. | |
Displays the first 10 bytes (including header) of selected VoFR subframes for the interface. |
This command is no longer supported in Cisco IOS Release 12.0(7)XK.
This command is no longer supported in Cisco IOS Release 12.0(7)XK.
This command is no longer supported in Cisco IOS Release 12.0(7)XK.
This command is no longer supported in Cisco IOS Release 12.0(7)XK.
This command is no longer supported in Cisco IOS Release 12.0(7)XK.
This command is no longer supported in Cisco IOS Release 12.0(7)XK.
Use the debug vpm all command to enable all voice port module (VPM) debugging. Use the no form of this command to disable all VPM debugging.
debug vpm allThis command has no arguments or keywords.
VPM debugging is not enabled.
| Release | Modification |
|---|---|
11.3(1)T | This command was introduced for the Cisco 3600 series. |
12.0(7)XK | This command was updated for the Cisco 2600, 3600, and MC3810. |
Use the debug vpm all command to enable the complete set of VPM debugging commands: debug vpm dsp, debug vpm error, debug vpm port, debug vpm spi, and debug vpm trunk_sc.
Execution of no debug all will turn off all port level debugging. It is usually a good idea to turn off all debugging and then enter the debug commands you are interested in one by one. This will help to avoid confusion about which ports you are actually debugging.
For sample outputs, refer to the individual commands in this chapter.
| Command | Description |
debug vpm port | Limits the debug vpm all command to a specified port. |
show debug | Shows which debug commands are enabled. |
Enables the display of trunk conditioning supervisory component trace information. |
Use the debug vpm trunk_sc privileged EXEC command to enable the display of trunk conditioning supervisory component trace information. The no form of this command disables the display of this information.
debug vpm trunk_scThis command has no arguments or keywords.
Trunk conditioning supervisory component trace information is not displayed.
| Release | Modification |
|---|---|
12.0(7)XK | This command was introduced on the Cisco 2600, 3600, and MC3810 platforms. |
Use the debug vpm port command with the slot-number/subunit-number/port argument to limit the debug vpm trunk_sc debug output to a particular port. If you do not use the debug vpm port command, the debug vpm trunk_sc displays output for all ports.
Execution of no debug all will turn off all port level debugging. It is usually a good idea to turn off all debugging and then enter the debug commands you are interested in one by one. This will help to avoid confusion about which ports you are actually debugging.
The following example shows debug vpm trunk_sc messages for port 1/0/0 on a Cisco 2600 or 3600 series router:
router#debug vpm trunk_scrouter#debug vpm port 1/0/0
The following example shows debug vpm trunk_sc messages for port 1/1 on a Cisco MC3810:
router#debug vpm trunk_scrouter#debug vpm port 1/1
The following example turns off debug vpm trunk_sc debugging messages:
router# no debug vpm trunk_sc
| Command | Description |
debug vpm all | Enables all VPM debugging |
debug vpm port | Limits the debug vpm trunk_sc command to a specified port. |
show debug | Shows which debug commands are enabled. |
This command has no arguments or keywords.
Debugging for vtsp is not enabled.
| Release | Modification |
|---|---|
12.0(3)T | This command was introduced on the Cisco AS5300 platform. |
12.0(7)XK | This command was first supported on the Cisco 2600, 3600 and MC3810 platforms. |
The debug vtsp all command enables the following debug vtsp commands: debug vtsp session, debug vtsp error, and debug vtsp dsp. For more information or sample output, refer to the individual commands in this chapter.
Execution of no debug vtsp all will turn off all VTSP-level debugging. You should turn off all debugging and then enter the debug commands you are interested in one by one. This will help to avoid confusion about which ports you are actually debugging.
| Command | Description |
show debug | Shows which debug commands are enabled. |
debug vtsp port | Limits vtsp debug output to a specific voice port. |
This command has no arguments or keywords.
Debugging for vtsp dsp is not enabled.
| Release | Modification |
|---|---|
12.0(3)T | This command was introduced on the Cisco AS5300 platform. |
12.0(7)XK | This command was first supported on the Cisco 2600, 3600, and MC3810 platforms. |
ON AS5300 ACCESS SERVERS
The debug vtsp dsp command shows messages from the DSP on the VFC to the router; this command can be useful if you suspect that the VFC is not functional. It is a simple way to check if the VFC is responding to off-hook indications.
ON 2600, 3600, MC3810 PLATFORMS
The debug vtsp dsp command shows messages from the DSP to the router.
The following example shows the collection of DTMF digits from the DSP on a Cisco AS5300 access server.
*Nov 30 00:44:34.491: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT: digit=3 *Nov 30 00:44:36.267: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT: digit=1 *Nov 30 00:44:36.571: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT: digit=0 *Nov 30 00:44:36.711: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT: digit=0 *Nov 30 00:44:37.147: vtsp_process_dsp_message: MSG_TX_DTMF_DIGIT: digit=2
| Command | Description |
Enables all VPM debugging. | |
debug vtsp port | Limits vtsp debug output to a specific voice port. |
show debug | Shows which debug commands are enabled. |
Use the debug vtsp error command to display processing errors in the voice telephony service provider. Use the no form of this command to disable vtsp error debugging.
debug vtsp errorThis command has no arguments or keywords.
Debugging for vtsp errors is not enabled.
| Release | Modification |
|---|---|
12.0(7)XK | This command was first supported on the Cisco 2600, 3600 and MC3810 platforms. |
The debug vtsp error command can be used to check for mismatches in interface capabilities.
The following example shows sample output from the debug vtsp error command, in which a dialed number is not reachable because it is not configured.
router#deb vtsp error
Voice telephony call control error debugging is on
router#
*Mar 1 00:21:48.698:cc_api_call_setup_ind (vdbPtr=0x1575AB0,
callInfo={called=,called_oct3=0x81,calling=9999,calling_oct3=0x0,called_oct3a=0x0,
fdest=0 peer_tag=1},callID=0x15896A4)
*Mar 1 00:21:48.698:cc_api_call_setup_ind type 3 , prot 0
*Mar 1 00:21:48.706:cc_process_call_setup_ind (event=0x16AD0E0) handed call to app
"SESSION"
*Mar 1 00:21:48.706:sess_appl:ev(23=CC_EV_CALL_SETUP_IND), cid(15), disp(0)
*Mar 1 00:21:48.706:sess_appl:ev(SSA_EV_CALL_SETUP_IND), cid(15), disp(0)
*Mar 1 00:21:48.706:ccCallSetContext (callID=0xF, context=0x1632898)
*Mar 1 00:21:48.706:ccCallSetupAck (callID=0xF)
*Mar 1 00:21:48.706:ccGenerateTone (callID=0xF tone=8)
*Mar 1 00:21:49.710:cc_api_call_digit_begin (vdbPtr=0x1575AB0, callID=0xF, digit=5,
flags=0x1, timestamp=0xB1AE6BC4, expiration=0x0)
*Mar 1 00:21:49.710:sess_appl:ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(15), disp(0)
*Mar 1 00:21:49.710:cid(15)st(SSA_CS_MAPPING)ev(SSA_EV_DIGIT_BEGIN)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
*Mar 1 00:21:49.714:ssaIgnore cid(15), st(SSA_CS_MAPPING),oldst(0), ev(10)
*Mar 1 00:21:49.778:cc_api_call_digit (vdbPtr=0x1575AB0, callID=0xF, digit=5,
duration=4165,tag 0, callparty 0 )
*Mar 1 00:21:49.778:sess_appl:ev(9=CC_EV_CALL_DIGIT), cid(15), disp(0)
*Mar 1 00:21:49.778:cid(15)st(SSA_CS_MAPPING)ev(SSA_EV_CALL_DIGIT)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
*Mar 1 00:21:49.782:ssaDigit
*Mar 1 00:21:49.782:ssaDigit, callinfo , digit 5, tag 0,callparty 0
*Mar 1 00:21:49.782:ssaDigit, calling 9999,result 1
*Mar 1 00:21:49.915:cc_api_call_digit_begin (vdbPtr=0x1575AB0, callID=0xF, digit=5,
flags=0x1, timestamp=0xB1AF6B6C, expiration=0x0)
*Mar 1 00:21:49.915:sess_appl:ev(10=CC_EV_CALL_DIGIT_BEGIN), cid(15), disp(0)
*Mar 1 00:21:49.915:cid(15)st(SSA_CS_MAPPING)ev(SSA_EV_DIGIT_BEGIN)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
*Mar 1 00:21:49.915:ssaIgnore cid(15), st(SSA_CS_MAPPING),oldst(0), ev(10)
*Mar 1 00:21:49.999:cc_api_call_digit (vdbPtr=0x1575AB0, callID=0xF, digit=5,
duration=95,tag 0, callparty 0 )
*Mar 1 00:21:49.999:sess_appl:ev(9=CC_EV_CALL_DIGIT), cid(15), disp(0)
*Mar 1 00:21:50.003:cid(15)st(SSA_CS_MAPPING)ev(SSA_EV_CALL_DIGIT)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
*Mar 1 00:21:50.003:ssaDigit
*Mar 1 00:21:50.003:ssaDigit, callinfo , digit 55, tag 0,callparty 0
*Mar 1 00:21:50.003:ssaDigit, calling 9999,result -1
*Mar 1 00:21:50.003:ccCallDisconnect (callID=0xF, cause=0x1C tag=0x0)
*Mar 1 00:21:50.003:ccCallDisconnect (callID=0xF, cause=0x1C tag=0x0)
*Mar 1 00:21:50.007:vtsp_process_event():prev_state = 0.4 ,
state = S_WAIT_RELEASE_NC, event = E_CC_DISCONNECT
Invalid FSM Input on channel 1/1:15
*Mar 1 00:21:52.927:vtsp_process_event():prev_state = 0.7 ,
state = S_WAIT_RELEASE_RESP, event = E_TSP_CALL_FEATURE_IND
Invalid FSM Input on channel 1/1:15
*Mar 1 00:21:52.931:cc_api_call_disconnect_done(vdbPtr=0x1575AB0, callID=0xF, disp=0,
tag=0x0)
*Mar 1 00:21:52.931:sess_appl:ev(13=CC_EV_CALL_DISCONNECT_DONE), cid(15), disp(0)
*Mar 1 00:21:52.931:cid(15)st(SSA_CS_DISCONNECTING)ev(SSA_EV_CALL_DISCONNECT_DONE)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(0)
| Command | Description |
Enables all VPM debugging. | |
debug vtsp port | Limits vtsp debug output to a specific voice port. |
show debug | Shows which debug commands are enabled. |
To observe the behavior of the VTSP state machine on a specific voice port, use the debug vtsp port command. Use the no form of the command to turn off the debug function.
For Cisco 2600 and 3600 series with analog voice ports:
debug vtsp port slot/subunit/port
no debug vtsp port slot/subunit/port
For Cisco 2600 and 3600 series with digital voice ports (with T1 packet voice trunk network modules):
debug vtsp port slot/port:ds0-group
no debug vtsp port slot/port:ds0-group
For Cisco MC3810 series with analog voice ports:
debug vtsp port slot/port
no debug vtsp port slot/port
For Cisco MC3810 series with digital voice ports:
debug vtsp port slot/port
no debug vtsp port slot/ds0-group
For the Cisco 2600 and 3600 series with analog voice ports:
slot/subunit/port | Debugs the analog voice port you specify with the slot/subunit/port designation. slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform. subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. (The VIC fits into the voice network module.) port specifies an analog voice port number. Valid entries are 0 and 1. |
For the Cisco 2600 and 3600 series with digital voice ports:
slot/port:ds0-group | Debugs the digital voice port you specify with the slot/port:ds0-group designation. slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform. port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. (One VWIC fits in an NM.) ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For the Cisco MC3810 series with analog voice ports:
slot/port | Debugs the analog voice port you specify with the slot/port designation. slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810. port specifies an analog voice port number. Valid entries are 1 to 6. |
For the Cisco MC3810 series with digital voice ports:
slot:ds0-group | Debugs the digital voice port you specify with the slot:ds0-group designation. slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1). ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
Debug vtsp commands are not limited to a specific port.
| Release | Modification |
|---|---|
12.0(3)XG | This command was introduced on Cisco 2600 and 3600 series routers. |
12.0(3)T | This command was introduced on the Cisco AS5300 platform. |
12.0(7)XK | This command was first supported on the Cisco MC3810 series. |
Use this command to limit the debug output to a particular voice port. The debug output can be quite voluminous for a single channel. The entire vtsp debug output form a platform with 12 voice ports might create problems. Use this debug with any or all of the other debug modes.
Execution of no debug vtsp all will turn off all VTSP-level debugging. It is usually a good idea to turn off all debugging and then enter the debug commands you are interested in one by one. This will help to avoid confusion about which ports you are actually debugging.
The following example shows sample output from the debug vtsp port 1/1/0 command:
router# debug vtsp port 1/1/0
*Mar 1 03:17:33.691: vtsp_tsp_call_setup_ind (sdb=0x613FD514, tdm_info=0x0,
tsp_info=0x613FD438, calling_number= called_number= redirect_number=): peer_tag=1110
*Mar 1 03:17:33.691: vtsp_do_call_setup_ind
*Mar 1 03:17:33.691: dsp_close_voice_channel: [] packet_len=8 channel_id=1
packet_id=75
*Mar 1 03:17:33.691: dsp_open_voice_channel: [] packet_len=12
channel_id=1 packet_id=74 alaw_ulaw_select=0 transport_protocol=2
*Mar 1 03:17:33.695: dsp_set_playout_delay: [] packet_len=18
channel_id=1 packet_id=76 mode=1 initial=60 min=4 max=200 fax_nom=300
*Mar 1 03:17:33.695: dsp_echo_canceller_control: [] packet_len=10 channel_id=1
packet_id=66 flags=0x0
*Mar 1 03:17:33.695: dsp_set_gains: [] packet_len=12 channel_id=1 packet_id=91
in_gain=0 out_gain=65506
*Mar 1 03:17:33.695: dsp_vad_enable: [] packet_len=10 channel_id=1 packet_id=78
thresh=-38
*Mar 1 03:17:33.695: vtsp_process_event(): [, 0.S_SETUP_INDICATED, E_CC_PROCEEDING]
*Mar 1 03:17:33.699: vtsp_process_event(): [, 0.S_SETUP_INDICATED,
E_CC_BRIDGE]act_bridge
*Mar 1 03:17:33.699: vtsp_ring_noan_timer_start: 1185370
*Mar 1 03:17:33.699: vtsp_process_event(): [, 0.S_SETUP_INDICATED,
E_CC_CAPS_IND]act_caps_ind
*Mar 1 03:17:33.699: act_caps_ind: Encap 2, Vad 2, Codec 0x1000, CodecBytes 60,
FaxRate 2, FaxBytes 30,
Sub-channel 10, Bitmask 0x0 SignalType 2
*Mar 1 03:17:33.703: vtsp_process_event(): [, 0.S_SETUP_INDICATED,
E_CC_CAPS_ACK]act_caps_ack
*Mar 1 03:17:33.703: dsp_idle_mode: [] packet_len=8 channel_id=1 packet_id=68
*Mar 1 03:17:33.703: vtsp_process_event(): [, 0.S_SETUP_INDICATED,
E_CC_CONNECT]act_connect
*Mar 1 03:17:33.703: vtsp_ring_noan_timer_stop: 1185370
*Mar 1 03:17:33.911: vtsp_process_event(): [, 0.S_CONNECT, E_DSPRM_PEND_SUCCESS]
act_pend_codec_success
*Mar 1 03:17:33.911: dsp_close_voice_channel: [] packet_len=8 channel_id=1
packet_id=75
*Mar 1 03:17:33.911: dsp_open_voice_channel: [] packet_len=12 channel_id=1
packet_id=74 alaw_ulaw_select=0 transport_protocol=2
*Mar 1 03:17:33.911: dsp_set_playout_delay: [] packet_len=18 channel_id=1 packet_id=76
mode=1 initial=60 min=4 max=200 fax_nom=300
*Mar 1 03:17:33.911: dsp_echo_canceller_control: [] packet_len=10 channel_id=1
packet_id=66 flags=0x0
*Mar 1 03:17:33.911: dsp_set_gains: [] packet_len=12 channel_id=1 packet_id=91
in_gain=0 out_gain=65506
*Mar 1 03:17:33.911: dsp_vad_enable: [] packet_len=10 channel_id=1 packet_id=78
thresh=-38
*Mar 1 03:17:33.911: dsp_encap_config: [] packet_len=24 channel_id=1 packet_id=
92 TransportProtocol 3 SID_support=0 sequence_number=0 rotate_flag=0 header_bytes 0xA0
*Mar 1 03:17:33.915: dsp_voice_mode: [] packet_len=22 channel_id=1 packet_id=73
coding_type=14 voice_field_size=60 VAD_flag=1 echo_length=128
comfort_noise=1 fax_detect=1 digit_relay=0
| Command | Description |
Enables all VPM debugging. | |
show debug | Shows which debug commands are enabled. |
This command has no arguments or keywords.
Debugging for vtsp session is not enabled.
| Release | Modification |
|---|---|
12.0(3)T | This command was introduced on the Cisco AS5300 platform. |
12.0(7)XK | This command was first supported on the Cisco 2600, 3600 and MC3810 platforms. |
The debug vtsp session command traces how the router interacts with the DSP based on the signaling indications from the signaling stack and requests from the application. This debug command displays information about how each network indication and application request is handled, signaling indications, and DSP control messages.
This debug level shows the internal workings of the voice telephony call state machine.
The following example shows sample output from the debug vtsp session command, in which the call has been accepted and the system is checking for incoming dial-peer matches:
*Nov 30 00:46:19.535: vtsp_tsp_call_accept_check (sdb=0x60CD4C58, calling_number=408 called_number=1): peer_tag=0 *Nov 30 00:46:19.535: vtsp_tsp_call_setup_ind (sdb=0x60CD4C58, tdm_info=0x60B80044, tsp_info=0x60B09EB0, calling_number=408 called_number=1): peer_tag=1
The following example shows sample output from the debug vtsp session command, in which a DSP has been allocated to handle the call and has indicated the call to the higher layer code:
*Nov 30 00:46:19.535: vtsp_do_call_setup_ind: *Nov 30 00:46:19.535: dsp_open_voice_channel: [0:D:12] packet_len=12 channel_id=8737 packet_id=74 alaw_ulaw_select=0 transport_protocol=2 *Nov 30 00:46:19.535: dsp_set_playout_delay: [0:D:12] packet_len=18 channel_id=8737 packet_id=76 mode=1 initial=60 min=4 max=200 fax_nom=300 *Nov 30 00:46:19.535: dsp_echo_canceller_control: [0:D:12] packet_len=10 channel_id=8737 packet_id=66 flags=0x0 *Nov 30 00:46:19.539: dsp_set_gains: [0:D:12] packet_len=12 channel_id=8737 packet_id=91 in_gain=0 out_gain=0 *Nov 30 00:46:19.539: dsp_vad_enable: [0:D:12] packet_len=10 channel_id=8737 packet_id=78 thresh=-38 *Nov 30 00:46:19.559: vtsp_process_event: [0:D:12, 0.3, 13] act_setup_ind_ack
The following example shows sample output from the debug vtsp session command, in which the higher layer code has accepted the call, placed the DSP in DTMF mode, and collected digits:
*Nov 30 00:46:19.559: dsp_voice_mode: [0:D:12] packet_len=20 channel_id=8737 packet_id=73 coding_type=1 voice_field_size=160 VAD_flag=0 echo_length=64 comfort_noise=1 fax_detect=1 *Nov 30 00:46:19.559: dsp_dtmf_mode: [0:D:12] packet_len=10 channel_id=8737 packet_id=65 dtmf_or_mf=0 *Nov 30 00:46:19.559: dsp_cp_tone_on: [0:D:12] packet_len=30 channel_id=8737 packet_id=72 tone_id=3 n_freq=2 freq_of_first=350 freq_of_second=440 amp_of_first=4000 amp_of_second=4000 direction=1 on_time_first=65535 off_time_first=0 on_time_second=65535 off_time_second=0 *Nov 30 00:46:19.559: vtsp_timer: 278792 *Nov 30 00:46:22.059: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:22.059: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:22.059: vtsp_timer: 279042 *Nov 30 00:46:22.363: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:22.363: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:22.363: vtsp_timer: 279072 *Nov 30 00:46:22.639: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:22.639: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:22.639: vtsp_timer: 279100 *Nov 30 00:46:22.843: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:22.843: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:22.843: vtsp_timer: 279120 *Nov 30 00:46:23.663: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:23.663: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:23.663: vtsp_timer: 279202
The following example shows sample output from the debug vtsp session command, in which the call proceeded and DTMF was disabled:
*Nov 30 00:46:23.663: vtsp_process_event: [0:D:12, 0.4, 15] act_dcollect_proc *Nov 30 00:46:23.663: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:23.663: dsp_idle_mode: [0:D:12] packet_len=8 channel_id=8737 packet_id=68
The following example shows sample output from the debug vtsp session command, in which the telephony call leg was conferenced with the packet network call leg, and the telephony call leg has performed capabilities exchange with the network-side call leg:
*Nov 30 00:46:23.699: vtsp_process_event: [0:D:12, 0.5, 17] act_bridge *Nov 30 00:46:23.699: vtsp_process_event: [0:D:12, 0.5, 22] act_caps_ind *Nov 30 00:46:23.699: vtsp_process_event: [0:D:12, 0.5, 23] act_caps_ack Go into voice mode with codec indicated in caps exchange. *Nov 30 00:46:23.699: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:23.699: dsp_idle_mode: [0:D:12] packet_len=8 channel_id=8737 packet_id=68 *Nov 30 00:46:23.699: dsp_voice_mode: [0:D:12] packet_len=20 channel_id=8737 packet_id=73 coding_type=6 voice_field_size=20 VAD_flag=1 echo_length=64 comfort_noise=1 fax_detect=1
The following example shows sample output from the debug vtsp session command in which the call has been connected at remote end:
*Nov 30 00:46:23.779: vtsp_process_event: [0:D:12, 0.5, 10] act_connect
The following example shows sample output from the debug vtsp session command in which disconnect was indicated and passed to upper layer:
*Nov 30 00:46:30.267: vtsp_process_event: [0:D:12, 0.11, 5] act_generate_disc
The following example shows sample output from the debug vtsp session command, in which the conference was torn down and the disconnect handshake was completed:
*Nov 30 00:46:30.267: vtsp_process_event: [0:D:12, 0.11, 18] act_bdrop *Nov 30 00:46:30.267: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:30.267: vtsp_process_event: [0:D:12, 0.11, 20] act_disconnect *Nov 30 00:46:30.267: dsp_get_error_stat: [0:D:12] packet_len=10 channel_id=0 packet_id=6 reset_flag=1 *Nov 30 00:46:30.267: vtsp_timer: 279862
The following example shows sample output from the debug vtsp session command, in which the final DSP statistics were retrieved:
*Nov 30 00:46:30.275: vtsp_process_event: [0:D:12, 0.17, 30] act_get_error *Nov 30 00:46:30.275: 0:D:12: rx_dropped=0 tx_dropped=0 rx_control=353 tx_control=338 tx_control_dropped=0 dsp_mode_channel_1=2 dsp_mode_channel_2=0 c[0]=71 c[1]=71 c[2]=71 c[3]=71 c[4]=68 c[5]=71 c[6]=68 c[7]=73 c[8]=83 c[9]=84 c[10]=87 c[11]=83 c[12]=84 c[13]=87 c[14]=71 c[15]=6 *Nov 30 00:46:30.275: dsp_get_levels: [0:D:12] packet_len=8 channel_id=8737 packet_id=89 *Nov 30 00:46:30.279: vtsp_process_event: [0:D:12, 0.17, 34] act_get_levels *Nov 30 00:46:30.279: dsp_get_tx_stats: [0:D:12] packet_len=10 channel_id=8737 packet_id=86 reset_flag=1 *Nov 30 00:46:30.287: vtsp_process_event: [0:D:12, 0.17, 31] act_stats_complete *Nov 30 00:46:30.287: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:30.287: dsp_idle_mode: [0:D:12] packet_len=8 channel_id=8737 packet_id=68 *Nov 30 00:46:30.287: vtsp_timer: 279864
The following example shows sample output from the debug vtsp session command, in which the DSP channel was closed and released:
*Nov 30 00:46:30.287: vtsp_process_event: [0:D:12, 0.18, 6] act_wrelease_release *Nov 30 00:46:30.287: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:30.287: dsp_idle_mode: [0:D:12] packet_len=8 channel_id=8737 packet_id=68 *Nov 30 00:46:30.287: dsp_close_voice_channel: [0:D:12] packet_len=8 channel_id=8737 packet_id=75 *Nov 30 00:46:30.287: vtsp_process_event: [0:D:12, 0.16, 42] act_terminate
| Command | Description |
Enables all VPM debugging. | |
debug vtsp port | Limits vtsp debug output to a specific voice port. |
show debug | Shows which debug commands are enabled. |
This command has no arguments or keywords.
Debugging for vtsp stats is not enabled.
| Release | Modification |
|---|---|
12.0(3)T | This command was introduced on the Cisco AS5300 platform. |
12.0(7)XK | This command was first supported on the Cisco 2600, 3600 and MC3810 platforms. |
The debug vtsp stats command generates a collection of DSP statistics for generating RTCP packets and a collection of other statistical information.
The following example shows sample debug vtsp stats output:
*Nov 30 00:53:26.499: vtsp_process_event: [0:D:14, 0.11, 19] act_packet_stats *Nov 30 00:53:26.499: dsp_get_voice_playout_delay_stats: [0:D:14] packet_len=10 channel_id=8753 packet_id=83 reset_flag=0 *Nov 30 00:53:26.499: dsp_get_voice_playout_error_stats: [0:D:14] packet_len=10 channel_id=8753 packet_id=84 reset_flag=0 *Nov 30 00:53:26.499: dsp_get_rx_stats: [0:D:14] packet_len=10 channel_id=8753 packet_id=87 reset_flag=0 *Nov 30 00:53:26.503: vtsp_process_dsp_message: MSG_TX_GET_VOICE_PLAYOUT_DELAY: clock_offset=-1664482334 curr_rx_delay_estimate=69 low_water_mark_rx_delay=69 high_water_mark_rx_delay=70 *Nov 30 00:53:26.503: vtsp_process_event: [0:D:14, 0.11, 28] act_packet_stats_res *Nov 30 00:53:26.503: vtsp_process_dsp_message: MSG_TX_GET_VOICE_PLAYOUT_ERROR: predective_concelement_duration=0 interpolative_concelement_duration=0 silence_concelement_duration=0 retroactive_mem_update=0 buf_overflow_discard_duration=10 num_talkspurt_detection_errors=0 *Nov 30 00:53:26.503: vtsp_process_event: [0:D:14, 0.11, 29] act_packet_stats_res *Nov 30 00:53:26.503: vtsp_process_dsp_message: MSG_TX_GET_RX_STAT: num_rx_pkts=152 num_early_pkts=-2074277660 num_late_pkts=327892 num_signalling_pkts=0 num_comfort_noise_pkts=0 receive_durtation=3130 voice_receive_duration=2970 fax_receive_duration=0 num_pack_ooseq=0 num_bad_header=0 *Nov 30 00:53:26.503: vtsp_process_event: [0:D:14, 0.11, 32] act_packet_stats_res
| Command | Description |
Enables all VPM debugging. | |
debug vtsp port | Limits vtsp debug output to a specific voice port. |
show debug | Shows which debug commands are enabled. |
To display the first 10 bytes (including header) of selected VoFR subframes for the interface, use the debug vtsp vofr subframe command in privileged EXEC mode. Use the no form of this command to turn off the debug function.
debug vtsp vofr subframe payload [from-dsp] [to-dsp]
no debug vtsp vofr subframe
payload | Number used to selectively display subframes of a specific payload. The payload types are: 0: Primary Payload - WARNING! This option may cause network instability. |
from-dsp | (Optional) Displays only the subframes received from the DSP. |
to-dsp | (Optional) Displays only the subframes going to the DSP. |
Privileged EXEC
| Release | Modification |
|---|---|
12.0(3)XG | This command was introduced on the Cisco 2600 and 3600. |
12.0(7)XK | Support for this command was extended to the Cisco MC3810. |
Each debug output displays the first 10 bytes of the FRF.11 subframe, including header bytes. Use the from-dsp and to-dsp options to limit the debugs to a single direction. If not specified, debugs are displayed for subframes when they are received from the DSP and before they are sent to the DSP.
Use extreme caution in selecting payload options 0 and 5. These options may cause network instability.
The following example shows sample output from the debug vtsp vofr subframe command:
router# debug vtsp vofr subframe 2
vtsp VoFR subframe debugging is enabled for payload 2 to and from DSP 3620_vofr#
*Mar 6 18:21:17.413:VoFR frame received from Network (24 bytes):9E 02 19 AA AA AA AA
AA AA AA
*Mar 6 18:21:17.449:VoFR frame received from DSP (18 bytes):9E 02 19 AA AA AA AA AA AA
AA
*Mar 6 18:21:23.969:VoFR frame received from Network (24 bytes):9E 02 19 AA AA AA AA
AA AA AA
*Mar 6 18:21:24.005:VoFR frame received from DSP (18 bytes):9E 02 19 AA AA AA AA AA AA
AA
| Command | Description |
debug vtsp all | Enables debugging of all VTSP areas. |
To observe the behavior of the VTSP state machine on a specific voice port, use the debug vtsp port command. Use the no form of the command to turn off the debug function.
For Cisco 2600 and 3600 series with analog voice ports:
debug vtsp port slot/subunit/port
no debug vtsp port slot/subunit/port
For Cisco 2600 and 3600 series with digital voice ports (with T1 packet voice trunk network modules):
debug vtsp port slot/port:ds0-group
no debug vtsp port slot/port:ds0-group
For Cisco MC3810 series with analog voice ports:
debug vtsp port slot/port
no debug vtsp port slot/port
For Cisco MC3810 series with digital voice ports:
debug vtsp port slot/port
no debug vtsp port slot/ds0-group
For the Cisco 2600 and 3600 series with analog voice ports:
slot/subunit/port | Debugs the analog voice port you specify with the slot/subunit/port designation. slot specifies a router slot in which a voice network module (NM) is installed. Valid entries are router slot numbers for the particular platform. subunit specifies a voice interface card (VIC) where the voice port is located. Valid entries are 0 and 1. (The VIC fits into the voice network module.) port specifies an analog voice port number. Valid entries are 0 and 1. |
For the Cisco 2600 and 3600 series with digital voice ports:
slot/port:ds0-group | Debugs the digital voice port you specify with the slot/port:ds0-group designation. slot specifies a router slot in which the packet voice trunk network module (NM) is installed. Valid entries are router slot numbers for the particular platform. port specifies a T1 or E1 physical port in the voice WAN interface card (VWIC). Valid entries are 0 and 1. (One VWIC fits in an NM.) ds0-group specifies a T1 or E1 logical port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
For the Cisco MC3810 series with analog voice ports:
slot/port | Debugs the analog voice port you specify with the slot/port designation. slot is the physical slot in which the analog voice module (AVM) is installed. The slot is always 1 for analog voice ports in the Cisco MC3810. port specifies an analog voice port number. Valid entries are 1 to 6. |
For the Cisco MC3810 series with digital voice ports:
slot:ds0-group | Debugs the digital voice port you specify with the slot:ds0-group designation. slot specifies the module (and controller). Valid entries are 0 for the MFT (controller 0) and 1 for the DVM (controller 1). ds0-group specifies a T1 or E1 logical voice port number. Valid entries are 0 to 23 for T1 and 0 to 30 for E1. |
Debug vtsp commands are not limited to a specific port.
| Release | Modification |
|---|---|
12.0(3)XG | This command was introduced on Cisco 2600 and 3600 series routers. |
12.0(3)T | This command was introduced on the Cisco AS5300 platform. |
12.0(7)XK | This command was first supported on the Cisco MC3810 series. |
Use this command to limit the debug output to a particular voice port. The debug output can be quite voluminous for a single channel. The entire vtsp debug output form a platform with 12 voice ports might create problems. Use this debug with any or all of the other debug modes.
Execution of no debug vtsp all will turn off all VTSP-level debugging. It is usually a good idea to turn off all debugging and then enter the debug commands you are interested in one by one. This will help to avoid confusion about which ports you are actually debugging.
The following example shows sample output from the debug vtsp port 1/1/0 command:
router# debug vtsp port 1/1/0
*Mar 1 03:17:33.691: vtsp_tsp_call_setup_ind (sdb=0x613FD514, tdm_info=0x0,
tsp_info=0x613FD438, calling_number= called_number= redirect_number=): peer_tag=1110
*Mar 1 03:17:33.691: vtsp_do_call_setup_ind
*Mar 1 03:17:33.691: dsp_close_voice_channel: [] packet_len=8 channel_id=1
packet_id=75
*Mar 1 03:17:33.691: dsp_open_voice_channel: [] packet_len=12
channel_id=1 packet_id=74 alaw_ulaw_select=0 transport_protocol=2
*Mar 1 03:17:33.695: dsp_set_playout_delay: [] packet_len=18
channel_id=1 packet_id=76 mode=1 initial=60 min=4 max=200 fax_nom=300
*Mar 1 03:17:33.695: dsp_echo_canceller_control: [] packet_len=10 channel_id=1
packet_id=66 flags=0x0
*Mar 1 03:17:33.695: dsp_set_gains: [] packet_len=12 channel_id=1 packet_id=91
in_gain=0 out_gain=65506
*Mar 1 03:17:33.695: dsp_vad_enable: [] packet_len=10 channel_id=1 packet_id=78
thresh=-38
*Mar 1 03:17:33.695: vtsp_process_event(): [, 0.S_SETUP_INDICATED, E_CC_PROCEEDING]
*Mar 1 03:17:33.699: vtsp_process_event(): [, 0.S_SETUP_INDICATED,
E_CC_BRIDGE]act_bridge
*Mar 1 03:17:33.699: vtsp_ring_noan_timer_start: 1185370
*Mar 1 03:17:33.699: vtsp_process_event(): [, 0.S_SETUP_INDICATED,
E_CC_CAPS_IND]act_caps_ind
*Mar 1 03:17:33.699: act_caps_ind: Encap 2, Vad 2, Codec 0x1000, CodecBytes 60,
FaxRate 2, FaxBytes 30,
Sub-channel 10, Bitmask 0x0 SignalType 2
*Mar 1 03:17:33.703: vtsp_process_event(): [, 0.S_SETUP_INDICATED,
E_CC_CAPS_ACK]act_caps_ack
*Mar 1 03:17:33.703: dsp_idle_mode: [] packet_len=8 channel_id=1 packet_id=68
*Mar 1 03:17:33.703: vtsp_process_event(): [, 0.S_SETUP_INDICATED,
E_CC_CONNECT]act_connect
*Mar 1 03:17:33.703: vtsp_ring_noan_timer_stop: 1185370
*Mar 1 03:17:33.911: vtsp_process_event(): [, 0.S_CONNECT, E_DSPRM_PEND_SUCCESS]
act_pend_codec_success
*Mar 1 03:17:33.911: dsp_close_voice_channel: [] packet_len=8 channel_id=1
packet_id=75
*Mar 1 03:17:33.911: dsp_open_voice_channel: [] packet_len=12 channel_id=1
packet_id=74 alaw_ulaw_select=0 transport_protocol=2
*Mar 1 03:17:33.911: dsp_set_playout_delay: [] packet_len=18 channel_id=1 packet_id=76
mode=1 initial=60 min=4 max=200 fax_nom=300
*Mar 1 03:17:33.911: dsp_echo_canceller_control: [] packet_len=10 channel_id=1
packet_id=66 flags=0x0
*Mar 1 03:17:33.911: dsp_set_gains: [] packet_len=12 channel_id=1 packet_id=91
in_gain=0 out_gain=65506
*Mar 1 03:17:33.911: dsp_vad_enable: [] packet_len=10 channel_id=1 packet_id=78
thresh=-38
*Mar 1 03:17:33.911: dsp_encap_config: [] packet_len=24 channel_id=1 packet_id=
92 TransportProtocol 3 SID_support=0 sequence_number=0 rotate_flag=0 header_bytes 0xA0
*Mar 1 03:17:33.915: dsp_voice_mode: [] packet_len=22 channel_id=1 packet_id=73
coding_type=14 voice_field_size=60 VAD_flag=1 echo_length=128
comfort_noise=1 fax_detect=1 digit_relay=0
| Command | Description |
Enables all VPM debugging. | |
show debug | Shows which debug commands are enabled. |
This command has no arguments or keywords.
Debugging for vtsp session is not enabled.
| Release | Modification |
|---|---|
12.0(3)T | This command was introduced on the Cisco AS5300 platform. |
12.0(7)XK | This command was first supported on the Cisco 2600, 3600 and MC3810 platforms. |
The debug vtsp session command traces how the router interacts with the DSP based on the signaling indications from the signaling stack and requests from the application. This debug command displays information about how each network indication and application request is handled, signaling indications, and DSP control messages.
This debug level shows the internal workings of the voice telephony call state machine.
The following example shows sample output from the debug vtsp session command, in which the call has been accepted and the system is checking for incoming dial-peer matches:
*Nov 30 00:46:19.535: vtsp_tsp_call_accept_check (sdb=0x60CD4C58, calling_number=408 called_number=1): peer_tag=0 *Nov 30 00:46:19.535: vtsp_tsp_call_setup_ind (sdb=0x60CD4C58, tdm_info=0x60B80044, tsp_info=0x60B09EB0, calling_number=408 called_number=1): peer_tag=1
The following example shows sample output from the debug vtsp session command, in which a DSP has been allocated to handle the call and has indicated the call to the higher layer code:
*Nov 30 00:46:19.535: vtsp_do_call_setup_ind: *Nov 30 00:46:19.535: dsp_open_voice_channel: [0:D:12] packet_len=12 channel_id=8737 packet_id=74 alaw_ulaw_select=0 transport_protocol=2 *Nov 30 00:46:19.535: dsp_set_playout_delay: [0:D:12] packet_len=18 channel_id=8737 packet_id=76 mode=1 initial=60 min=4 max=200 fax_nom=300 *Nov 30 00:46:19.535: dsp_echo_canceller_control: [0:D:12] packet_len=10 channel_id=8737 packet_id=66 flags=0x0 *Nov 30 00:46:19.539: dsp_set_gains: [0:D:12] packet_len=12 channel_id=8737 packet_id=91 in_gain=0 out_gain=0 *Nov 30 00:46:19.539: dsp_vad_enable: [0:D:12] packet_len=10 channel_id=8737 packet_id=78 thresh=-38 *Nov 30 00:46:19.559: vtsp_process_event: [0:D:12, 0.3, 13] act_setup_ind_ack
The following example shows sample output from the debug vtsp session command, in which the higher layer code has accepted the call, placed the DSP in DTMF mode, and collected digits:
*Nov 30 00:46:19.559: dsp_voice_mode: [0:D:12] packet_len=20 channel_id=8737 packet_id=73 coding_type=1 voice_field_size=160 VAD_flag=0 echo_length=64 comfort_noise=1 fax_detect=1 *Nov 30 00:46:19.559: dsp_dtmf_mode: [0:D:12] packet_len=10 channel_id=8737 packet_id=65 dtmf_or_mf=0 *Nov 30 00:46:19.559: dsp_cp_tone_on: [0:D:12] packet_len=30 channel_id=8737 packet_id=72 tone_id=3 n_freq=2 freq_of_first=350 freq_of_second=440 amp_of_first=4000 amp_of_second=4000 direction=1 on_time_first=65535 off_time_first=0 on_time_second=65535 off_time_second=0 *Nov 30 00:46:19.559: vtsp_timer: 278792 *Nov 30 00:46:22.059: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:22.059: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:22.059: vtsp_timer: 279042 *Nov 30 00:46:22.363: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:22.363: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:22.363: vtsp_timer: 279072 *Nov 30 00:46:22.639: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:22.639: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:22.639: vtsp_timer: 279100 *Nov 30 00:46:22.843: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:22.843: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:22.843: vtsp_timer: 279120 *Nov 30 00:46:23.663: vtsp_process_event: [0:D:12, 0.4, 25] act_dcollect_digit *Nov 30 00:46:23.663: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:23.663: vtsp_timer: 279202
The following example shows sample output from the debug vtsp session command, in which the call proceeded and DTMF was disabled:
*Nov 30 00:46:23.663: vtsp_process_event: [0:D:12, 0.4, 15] act_dcollect_proc *Nov 30 00:46:23.663: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:23.663: dsp_idle_mode: [0:D:12] packet_len=8 channel_id=8737 packet_id=68
The following example shows sample output from the debug vtsp session command, in which the telephony call leg was conferenced with the packet network call leg, and the telephony call leg has performed capabilities exchange with the network-side call leg:
*Nov 30 00:46:23.699: vtsp_process_event: [0:D:12, 0.5, 17] act_bridge *Nov 30 00:46:23.699: vtsp_process_event: [0:D:12, 0.5, 22] act_caps_ind *Nov 30 00:46:23.699: vtsp_process_event: [0:D:12, 0.5, 23] act_caps_ack Go into voice mode with codec indicated in caps exchange. *Nov 30 00:46:23.699: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:23.699: dsp_idle_mode: [0:D:12] packet_len=8 channel_id=8737 packet_id=68 *Nov 30 00:46:23.699: dsp_voice_mode: [0:D:12] packet_len=20 channel_id=8737 packet_id=73 coding_type=6 voice_field_size=20 VAD_flag=1 echo_length=64 comfort_noise=1 fax_detect=1
The following example shows sample output from the debug vtsp session command in which the call has been connected at remote end:
*Nov 30 00:46:23.779: vtsp_process_event: [0:D:12, 0.5, 10] act_connect
The following example shows sample output from the debug vtsp session command in which disconnect was indicated and passed to upper layer:
*Nov 30 00:46:30.267: vtsp_process_event: [0:D:12, 0.11, 5] act_generate_disc
The following example shows sample output from the debug vtsp session command, in which the conference was torn down and the disconnect handshake was completed:
*Nov 30 00:46:30.267: vtsp_process_event: [0:D:12, 0.11, 18] act_bdrop *Nov 30 00:46:30.267: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:30.267: vtsp_process_event: [0:D:12, 0.11, 20] act_disconnect *Nov 30 00:46:30.267: dsp_get_error_stat: [0:D:12] packet_len=10 channel_id=0 packet_id=6 reset_flag=1 *Nov 30 00:46:30.267: vtsp_timer: 279862
The following example shows sample output from the debug vtsp session command, in which the final DSP statistics were retrieved:
*Nov 30 00:46:30.275: vtsp_process_event: [0:D:12, 0.17, 30] act_get_error *Nov 30 00:46:30.275: 0:D:12: rx_dropped=0 tx_dropped=0 rx_control=353 tx_control=338 tx_control_dropped=0 dsp_mode_channel_1=2 dsp_mode_channel_2=0 c[0]=71 c[1]=71 c[2]=71 c[3]=71 c[4]=68 c[5]=71 c[6]=68 c[7]=73 c[8]=83 c[9]=84 c[10]=87 c[11]=83 c[12]=84 c[13]=87 c[14]=71 c[15]=6 *Nov 30 00:46:30.275: dsp_get_levels: [0:D:12] packet_len=8 channel_id=8737 packet_id=89 *Nov 30 00:46:30.279: vtsp_process_event: [0:D:12, 0.17, 34] act_get_levels *Nov 30 00:46:30.279: dsp_get_tx_stats: [0:D:12] packet_len=10 channel_id=8737 packet_id=86 reset_flag=1 *Nov 30 00:46:30.287: vtsp_process_event: [0:D:12, 0.17, 31] act_stats_complete *Nov 30 00:46:30.287: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:30.287: dsp_idle_mode: [0:D:12] packet_len=8 channel_id=8737 packet_id=68 *Nov 30 00:46:30.287: vtsp_timer: 279864
The following example shows sample output from the debug vtsp session command, in which the DSP channel was closed and released:
*Nov 30 00:46:30.287: vtsp_process_event: [0:D:12, 0.18, 6] act_wrelease_release *Nov 30 00:46:30.287: dsp_cp_tone_off: [0:D:12] packet_len=8 channel_id=8737 packet_id=71 *Nov 30 00:46:30.287: dsp_idle_mode: [0:D:12] packet_len=8 channel_id=8737 packet_id=68 *Nov 30 00:46:30.287: dsp_close_voice_channel: [0:D:12] packet_len=8 channel_id=8737 packet_id=75 *Nov 30 00:46:30.287: vtsp_process_event: [0:D:12, 0.16, 42] act_terminate
| Command | Description |
Enables all VPM debugging. | |
debug vtsp port | Limits vtsp debug output to a specific voice port. |
show debug | Shows which debug commands are enabled. |
This command has no arguments or keywords.
Debugging for vtsp stats is not enabled.
| Release | Modification |
|---|---|
12.0(3)T | This command was introduced on the Cisco AS5300 platform. |
12.0(7)XK | This command was first supported on the Cisco 2600, 3600 and MC3810 platforms. |
The debug vtsp stats command generates a collection of DSP statistics for generating RTCP packets and a collection of other statistical information.
The following example shows sample debug vtsp stats output:
*Nov 30 00:53:26.499: vtsp_process_event: [0:D:14, 0.11, 19] act_packet_stats *Nov 30 00:53:26.499: dsp_get_voice_playout_delay_stats: [0:D:14] packet_len=10 channel_id=8753 packet_id=83 reset_flag=0 *Nov 30 00:53:26.499: dsp_get_voice_playout_error_stats: [0:D:14] packet_len=10 channel_id=8753 packet_id=84 reset_flag=0 *Nov 30 00:53:26.499: dsp_get_rx_stats: [0:D:14] packet_len=10 channel_id=8753 packet_id=87 reset_flag=0 *Nov 30 00:53:26.503: vtsp_process_dsp_message: MSG_TX_GET_VOICE_PLAYOUT_DELAY: clock_offset=-1664482334 curr_rx_delay_estimate=69 low_water_mark_rx_delay=69 high_water_mark_rx_delay=70 *Nov 30 00:53:26.503: vtsp_process_event: [0:D:14, 0.11, 28] act_packet_stats_res *Nov 30 00:53:26.503: vtsp_process_dsp_message: MSG_TX_GET_VOICE_PLAYOUT_ERROR: predective_concelement_duration=0 interpolative_concelement_duration=0 silence_concelement_duration=0 retroactive_mem_update=0 buf_overflow_discard_duration=10 num_talkspurt_detection_errors=0 *Nov 30 00:53:26.503: vtsp_process_event: [0:D:14, 0.11, 29] act_packet_stats_res *Nov 30 00:53:26.503: vtsp_process_dsp_message: MSG_TX_GET_RX_STAT: num_rx_pkts=152 num_early_pkts=-2074277660 num_late_pkts=327892 num_signalling_pkts=0 num_comfort_noise_pkts=0 receive_durtation=3130 voice_receive_duration=2970 fax_receive_duration=0 num_pack_ooseq=0 num_bad_header=0 *Nov 30 00:53:26.503: vtsp_process_event: [0:D:14, 0.11, 32] act_packet_stats_res
| Command | Description |
Enables all VPM debugging. | |
debug vtsp port | Limits vtsp debug output to a specific voice port. |
show debug | Shows which debug commands are enabled. |
To display the first 10 bytes (including header) of selected VoFR subframes for the interface, use the debug vtsp vofr subframe command. Use the no form of the command to turn off the debug function.
debug vtsp vofr subframe payload [from-dsp] [to-dsp]
no debug vtsp vofr subframe
payload | Number used to selectively display subframes of a specific payload. The payload types are: 0: Primary Payload - WARNING! This option may cause network instability |
from-dsp | Displays only the subframes received from the DSP. |
to-dsp | Displays only the subframes going to the DSP. |
Debugging for vtsp vofr subframe is not enabled.
| Release | Modification |
|---|---|
12.0(3)XG, 12.0(4)T | This command was introduced on the Cisco 2600 and 3600 platforms. |
12.0(7)XK | This command was first supported on the Cisco MC3810 platform. |
Each debug output displays the first 10 bytes of the FRF.11 subframe, including header bytes. The from-dsp and to-dsp options can be used to limit the debugs to a single direction. If not specified, debugs are displayed for subframes when they are received from the DSP and before they are sent to the DSP.
Use extreme caution in selecting payload options 0 and 6. These options may cause network instability.
The following example shows sample output from the debug vtsp vofr subframe command:
router# debug vtsp vofr subframe 2
vtsp VoFR subframe debugging is enabled for payload 2 to and from DSP 3620_vofr#
*Mar 6 18:21:17.413:VoFR frame received from Network (24 bytes):9E 02 19 AA AA AA AA
AA AA AA
*Mar 6 18:21:17.449:VoFR frame received from DSP (18 bytes):9E 02 19 AA AA AA AA AA AA
AA
*Mar 6 18:21:23.969:VoFR frame received from Network (24 bytes):9E 02 19 AA AA AA AA
AA AA AA
*Mar 6 18:21:24.005:VoFR frame received from DSP (18 bytes):9E 02 19 AA AA AA AA AA AA
AA
| Command | Description |
Enables all VPM debugging. | |
debug vtsp port | Limits vtsp debug output to a specific voice port. |
show debug | Shows which debug commands are enabled. |
Posted: Thu Sep 28 10:39:00 PDT 2000
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