Configuring Voice over IP for Cisco MC3810 Series Concentrators Feature Summary Voice over IP (VoIP) enables a Cisco MC3810 concentrator to carry voice traffic (for example, telephone calls and faxes) over an IP network. Voice over IP is primarily a software feature; however, to support this feature, a Cisco MC3810 must be equipped with a digital voice module (DVM) or an analog voice module (AVM).
Benefits • Transparent CCS and Frame Forwarding Enhancments on the Cisco MC3810, Cisco IOS Release 12.0(7)XK online document • Voice Port Enhancements on Cisco 2600 and 3600 Series Routers and MC3810 Concentrators, Cisco IOS Release 12.0(7)XK online document Supported Platform • Cisco MC3810 series concentrators Supported Standards, MIBs, and RFCs This feature supports the following standards and RFCs: • • • ITU-T H.
Benefits (c) Configuring IP RTP Priority Refer to the “Configuring IP Networks for Real-Time Voice Traffic” section for information about how to select and configure the appropriate QoS tools to optimize voice traffic on your network. 3 Configuring Number Expansion Use the num-exp command to configure number expansion if your telephone network is configured so that you can reach a destination by dialing only a portion (an extension number) of the full E.164 telephone number.
Preparing to Configure VoIP Under most circumstances, the default voice-port command values are adequate to configure FXO and FXS ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks, E&M ports might need specific voice-port values configured, depending on the specifications of the devices in your telephony network. 7 Configuring the H.323 Gateway The gateway capability allows a Cisco MC3810 to function as an H.323 endpoint.
Configuring Multilink PPP with Interleaving in your network do not necessarily perform the same operations; the QoS tasks they perform might differ as well. To configure your IP network for real-time voice traffic, you need to take into consideration the functions of both edge and backbone routers in your network, then select the appropriate QoS tool or tools.
Configuring IP Networks for Real-Time Voice Traffic Note Multilink PPP should not be used on links greater than 2 Mbps. Multilink PPP support for interleaving can be configured on virtual templates, dialer interfaces, and ISDN BRI or PRI interfaces. To configure interleaving, you need to complete the following tasks: • Configure the dialer interface or virtual template, as defined in the relevant chapters of the Cisco IOS 12.0 Dial Solutions Configuration Guide.
Configuring RTP Header Compression Figure 1 RTP Header Compression Before RTP header compression: 20 bytes IP 8 bytes 12 bytes UDP RTP Header Payload 20 to 160 bytes After RTP header compression: 2 to 4 bytes IP/UDP/RTP header 20 to 160 bytes 12076 Payload You should configure RTP header compression if the following conditions exist in your network: • • Slow links Need to save bandwidth Note RTP header compression should not be used on links greater than 2 Mbps.
Configuring IP Networks for Real-Time Voice Traffic Change the Number of Header Compression Connections By default, the software supports a total of 32 RTP header compression connections on an interface. To specify a different number of RTP header compression connections, use the following command in interface configuration mode: Command Purpose router(config-if)# ip rtp compression connections number Specify the total number of RTP header compression connections supported on an interface.
Configuring Number Expansion To reserve a strict priority queue for a set of RTP packet flows belonging to a range of UDP destination ports, use the following command in interface configuration mode: Command Purpose router(config-if)# ip rtp priority starting-rtp-port-number port-number-range bandwidth Reserves a strict priority queue for a set of RTP packet flows belonging to a range of UDP destination ports.
Configuring Dial Peers Step Command Purpose 2 router(config-dialpeer)# port slot/port Associate this POTS dial peer with a specific voice port. To configure direct inward dial (DID) for a particular POTS dial peer, use the following commands beginning in global configuration mode: Step Command Purpose 1 router(config)# dial-peer voice number pots Enter dial-peer configuration mode to configure a POTS peer.
Configuring Dial Peer Hunting Step Command Purpose 3 router(config-dialpeer)# dtmf-relay [cisco-rtp] [h245-signal] [h245-alphanumeric] (Optional) Specify how an H.323 gateway relays DTMF tones through an IP network. Options allow the gateway to forward tones “out-of-band”, or separate from the voice stream. Note This command is only supported if your Cisco MC3810 has version 549 or newer DSPs.
Configuring Dial Peers Step Command Purpose 1 router(config)# dial-peer hunt (Optional) Specify the hunting selection order for dial peers. 2 router(config)# dial-peer terminator character (Optional) Designate a terminating character for variable length dialed numbers. The default character is # (pound sign). If using dial peer hunting, there may be situations in which you want to disable dial-peer hunting on a specific dial peer.
Optimizing Dial Peer and Network Interface Configurations Optimizing Dial Peer and Network Interface Configurations Depending on how you have configured your network interfaces, you might need to configure additional VoIP dial-peer parameters.
Optimizing Dial Peer and Network Interface Configurations Step Command Purpose 2 router(config-dialpeer)# codec {g711alaw | g711ulaw | g723ar53 | g723ar63 | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g728 | g729abr8 | g729ar8 | g729br8 | g729r8}[bytes payload-size] Specify the desired voice coder rate of speech.Optionally specify the voice payload (in bytes) of each frame. The default for the codec command is g729r8; normally the default configuration for this command is the most desirable.
Configuring Codec Selection Order To configure a voice class in which you can define the order of preference in which a router selects a codec when it negotiates with a far-end router, enter the following commands beginning in global configuration mode: Step Command Purpose Create a voice class for a codec preference list. The range for the tag number is 1 to 10000. The tag number must be unique on the router.
Configuring Voice Ports Verifying Codec Settings of Dial Peers To display the codec voice-classes assigned to VoIP dial peers, enter the show running-config command. The following example shows exerpts from the show running-config command output, where three codec voice classes (10, 20 and 30) have been applied to three VoIP dial peers (101, 102 and 102): router# show running-config Building configuration... Current configuration: ! version 12.0 . . .
Configuring FXO or FXS Voice Ports • • • Configuring E&M Voice Ports Fine-Tuning E&M Voice Ports Activating the Voice Port Configuring FXO or FXS Voice Ports Under most circumstances the default values are adequate for FXO and FXS voice ports. Task List If you need to change the default configuration for these voice ports, perform the following tasks: 1 Configure the applicable parameters for the voice port. 2 Verify the configuration. 3 Troubleshoot and correct any configuration errors.
Configuring Voice Ports Step Command Purpose 7 router(config-voice-port)#compand-type {u-law | a-law} Configure the companding standard used to convert between analog and digital signals in PCM systems. Defaults are: u-law for T1; a-law for E1. 8 router(config-voice-port)#vad (Optional) Enable voice activity detection (VAD). 9 router(config-voice-port)#comfort-noise (Optional) Enable background noise if VAD is enabled.
Fine-Tuning FXO and FXS Voice Ports Fine-Tuning FXO and FXS Voice Ports Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation. The commands for these parameters are referred to as voice-port tuning commands. Note In most cases, the default values for voice-port tuning commands will be sufficient.
Configuring Voice Ports Step Command Purpose 8 router(config-voiceport)# 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} (For T1/E1 digital voice ports only.) Configure the voice port to manipulate the transmit and/or receive bit patterns to match the bit patterns required by a connected device. Be careful not to destroy the information content of the bit pattern.
Configuring E&M Voice Ports Step Command Purpose 22 router(config-voiceport)# disconnect-ack (FXS only) Configure the voice port to return an acknowledgment upon receipt of a disconnect signal. 23 router(config-voiceport)# ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07 | pattern08 | pattern09 | pattern10 | pattern11 | pattern12 ] [define pulse-interval]} (FXS only) Specify the on and off times for the ringing pulses.
Configuring Voice Ports Step Command Purpose 2 router(config-voiceport)# connection {plar | tie-line | trunk | plar-opx} destination-string [answer-mode] Specify the voice-port connection type and the destination telephone number. • 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.
Configuring E&M Voice Ports Step Command Purpose 6 router(config-voiceport)# type {1 | 2 | 3 | 5} Select the appropriate E&M interface type.
Configuring Voice Ports Validation Tips You can check the validity of your voice-port configuration by performing the following tasks: • • Pick up the handset of an attached telephony device and check for dial tone. • • • Use the show voice port command to view the voice-port configuration. If you have dial tone, check for DTMF detection. If the dial tone stops when you dial a digit, the voice port is most likely configured properly.
Fine-Tuning E&M Voice Ports Step Command Purpose 1 router# configure terminal Enter global configuration mode. 2 router(config)# voice-port slot/port Identify the voice port you want to configure and enter voice-port configuration mode. 3 router(config-voiceport)# input gain value Specify the receive gain (in dB) for the voice port. Value range is –6 to 14. 4 router(config-voiceport)# output attenuation value Specify the transmit attenuation (in dB) for the voice port. Value range is 0 to 14.
Configuring Voice Ports Step Command Purpose 16 router(config-voiceport)# timing clear-wait milliseconds Specify the number of milliseconds between the inactive seizure signal and the call being cleared. The range is 100 to 2000. The default is 400. 17 router(config-voiceport)# timing delay-duration milliseconds Specify the delay signal duration in milliseconds for delay dial signaling. This command applies only if the signal command is set to delay-dial. The range is 100 to 5000.
Activating the Voice Port Activating the Voice Port After you have configured the voice port, you need to activate the voice port to bring it online. Cisco recommends that you cycle the port—shut the port down and then bring it online again. To activate a voice port, enter the following command in voice-port configuration mode: Command Purpose router(config-voiceport)# no shutdown Activate the voice port.
Configuring the H.323 Gateway Enabling VoIP Gateway Functionality Enable VoIP gateway functionality by using the gateway command. To enable gateway functionality, use the following commands: Step Command Purpose 1 router# configure terminal Enter global configuration mode. 2 router(config)# gateway Enable the VoIP gateway. Configuring Gateway Interface Parameters The next step in configuring an H.323 gateway is to configure the gateway interface parameters.
Linking PBX Users with E&M Trunk Lines Configuration Example The actual Voice over IP configuration procedure you complete depends on the actual topology of your voice network. The following configuration examples should give you a starting point. Of course, these configuration examples would need to be customized to reflect your network topology.
Configuring the H.323 Gateway !Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern 555.... port 1/0/0 !Configure pots dial peer 2 dial-peer voice 2 pots destination-pattern 555.... port 1/0/1 !Configure voip dial peer 3 dial-peer voice 3 voip destination-pattern 119.... session target ipv4:172.16.65.
PSTN Gateway Access Using FXO Connection interface serial 0/0 description serial interface type dte ip address 172.16.65.182 no shutdown Note PBXs should be configured to pass all DTMF signals to the Cisco voice router. Cisco recommends that you do not configure store and forward tone. Note If you change the gain or the telephony port, make sure that the telephony port still accepts DTMF signals.
Configuring the H.323 Gateway ! Configure the serial interface interface serial 0/0 clock rate 2000000 ip address 172.16.1.123 no shutdown Configuration for Router SLC ! Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern 9........... port 1/0/0 ! Configure voip dial peer 2 dial-peer voice 2 voip destination-pattern +14085554000 session target ipv4:172.16.1.123 ! Configure serial interface interface serial 0/0 ip address 172.16.65.
Codec Preference Configuration Configuration for Router SJ ! Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern +14085554000 port 1/0/0 ! Configure voip dial peer 2 dial-peer voice 2 voip destination-pattern 9........... session target ipv4:172.16.65.182 ! Configure the serial interface interface serial 0/0 clock rate 2000000 ip address 172.16.1.123 no shutdown Configuration for Router SLC ! Configure pots dial peer 1 dial-peer voice 1 pots destination-pattern 9...........
Configuring the H.323 Gateway The following example assigns a voice class 10 to a VoIP dial peer: router(config)# dial-peer voice 25 voip router(config-dial-peer)# voice-class codec 10 34 Release 12.
Codec Preference Configuration Command Reference This section documents new or modified commands. Modified commands are indicated by an asterisk (*). All other commands used on these platforms are documented in the Cisco IOS Release 12.0 command reference publications.
codec preference codec preference To define the order of preference in which network dial peers select codecs, use the codec preference voice-class configuration command. Enter the no form of this command to restore the default order of preference. codec preference priority codec bytes payload-size no codec preference Syntax Description priority The order of selection preference you assign to a codec. The valid range is 1 to 12, where 1 is the highest priority. codec Codec options.
codec preference Command Modes Voice class configuration Command History Release Modification 12.0(2)XH This command was introduced on the Cisco AS5300. 12.0(7)T This command was first supported on the Cisco 2600 and 3600 series routers. 12.0(7)XK This command was first supported on the Cisco MC3810 series. Usage Guidelines The routers at opposite ends of the WAN may have to negotiate the codec selection for the network dial peers.
codec preference Table 1 Voice Payload-per-Frame Options and Defaults Codec Protocol Voice Payload Options (bytes) Default Voice Payload (bytes) g729abr8 g729ar8 g729br8 g729r8 VoIP VoFR VoATM 10 to 230 in multiples of 10 10 to 240 in multiples of 10 10 to 240 in multiples of 10 20 30 30 Examples The following example shows how to create a voice class and specify a codec selection preference for the voice class starting from global configuration mode: router(config)# voice router(config-class)# r
connection connection To specify a connection mode for a voice port, use the connection voice-port configuration command. Use the no form of this command to disable the selected connection mode. connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]} no connection {plar | tie-line | plar-opx} digits | {trunk digits [answer-mode]} Syntax Description plar Specifies a private line auto ring down (PLAR) connection.
connection Usage Guidelines Use this command to specify a connection mode for a specific interface. For example, use the connection plar command to specify a PLAR interface. The string you configure for this command is used as the called number for all incoming calls over this connection. The destination peer is determined by the called number. Use the connection trunk command to specify a straight tie-line connection to a PBX.
connection Related Commands Command Description destination-pattern Specifies either the prefix or the full E.164 telephone number to be used for a dial peer. dial-peer voice Enters dial-peer configuration mode and specifies the method of voice-related encapsulation. session-protocol Establishes a session protocol for calls between the local and remote routers via the packet network. session-target Configures a network-specific address for a dial peer.
dial-peer hunt dial-peer hunt To specify a hunt selection order for dial-peers, use the dial-peer hunt dial-peer configuration command. Use the no form of this command to restore the default selection order. dial-peer hunt hunt-order-number no dial-peer hunt Syntax Description 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.
dial-peer hunt configuration. “Least recent use” refers to the destination pattern that has waited the longest since being selected. “Random selection” weights all of the destination patterns equally in a random selection mode. Example The following example configures the dial peers to hunt in the following order: (1) longest match in phone number, (2) explicit preference, (3) random selection.
dial-peer terminator dial-peer terminator To change the character used as a terminator for variable length dialed numbers, use the dial-peer terminator global configuration command. Use the no form of this command to restore the default terminating character. dial-peer terminator character no dial-peer terminator Syntax Description character Designates the terminating character for a variable-length dialed number. Valid numbers and characters are #, *, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, a, b, c, and d.
dial-peer terminator Related Commands 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.
dial-peer voice dial-peer voice To enter dial-peer configuration mode and specify the method of voice encapsulation, use the dial-peer voice global configuration command. Use the no form of this command to disable the selected encapsulation mode.
dial-peer voice Command History Release Modification 11.3(1)T This command was first introduced. 11.3(1)MA This command was first supported on the Cisco MC3810, with support for POTS, VoFR, and VoATM. 12.0(3)XG and 12.0(4)T This command added VoFR to the Cisco 2600 and 3600 series routers. 12.0(4)T This command added VoFR to the Cisco 7200 series platform. 12.0(7)XK This command added VoIP to the Cisco MC3810 and VoATM to the Cisco 3600 series routers.
ds0-group ds0-group To specify the DS0 timeslots that make up a logical voice port on a T1 or E1 controller, and to specify the signaling type, use the ds0-group controller configuration command. Use the no form of the command to remove the DS0 group and signaling setting. ds0-group ds0-group-no timeslots timeslot-list type signal-type no ds0-group ds0-group-no Syntax Description ds0-group-no A value from 0 to 23 that identifies the DS0 group.
ds0-group The following options are available only on E1 controllers on the Cisco MC3810: • e&m-melcas-immed—E&M Mercury Exchange Limited Channel Associated Signaling (MELCAS) immediate start signaling support • e&m-melcas-wink—E&M MELCAS wink start signaling support • e&m-melcas-delay—E&M MELCAS delay start signaling support • fxo-melcas—MELCAS Foreign Exchange Office signaling support • fxs-melcas—MELCAS Foreign Exchange Station signaling support The following options are available only when the mode ccs
ds0-group On the Cisco MC3810, the slot number is the controller number. Although only one voice port is created for each group, applicable calls are routed to any channel in the group. On the Cisco MC3810 when configured for transparent CCS, the channel type configured as the ext-sig-master is considered the master side of the permanent virtual circuit (PVC) connection which is responsible for establishing the PVC connection. After the master channel is configured, a fixed timer of 30 seconds starts.
dtmf-relay dtmf-relay Use the dtmf-relay command to specify how an H.323 gateway relays DTMF tones through an IP network. Options allow the gateway to forward tones “out-of-band”, or separate from the voice stream. The no form of this command removes all signaling options and transmits the DTMF tones as part of the audio stream.
dtmf-relay Note The cisco-rtp version of dtmf-relay is a proprietary Cisco implementation and only interoperates between Cisco AS5300 universal access servers, Cisco 2600 or 3600 modular access routers, or Cisco MC3810 concentrators running Cisco IOS Release 12.0(7)XK, or later releases. Otherwise, the DTMF relay feature will not function and the gateway will send DTMF tones inband.
forward-digits forward-digits To specify which digits to forward for voice calls, use the forward-digits dial-peer configuration command. If the no form of this command is entered, any digits not matching the destination-pattern are not forwarded. Use the default form of this command to restore the default state. forward-digits {num-digit | all | extra} no forward-digits default forward-digits Syntax Description num-digit The number of digits to be forwarded.
forward-digits Use the default form of this command if a non-default digit-forwarding scheme was entered previously, and you wish to restore the default. For QSIG ISDN connections, entering forward-digits all implies that all of the digits of the called party number are sent to the ISDN connection. When you enter forward-digits num-digit and enter a number from 1 to 32, the number of digits specified (right justified) of the called part number are sent to the ISDN connection.
huntstop huntstop To disable all further dial-peer hunting if a call fails when using hunt groups, enter the huntstop dial-peer configuration command. To reenable dial-peer call hunting, enter the no form of this command. huntstop no huntstop Syntax Description This command has no arguments or keywords. Defaults Disabled Command Modes Dial-peer configuration Command History Release Modification 12.0(5)T This command was introduced on the Cisco MC3810. 12.
icpif icpif To specify the Impairment/Calculated Planning Impairment Factor (ICPIF) for calls sent by a dial peer, use the icpif dial peer configuration command. Use the no form of this command to restore the default value for this command. icpif number no icpif number Syntax Description number Integer, expressed in equipment impairment factor units, specifying the ICPIF value. Valid entries are from 0 to 55. Default The default value for this command is 30.
incoming called-number incoming called-number To identify the service type for a call on a router handling both voice and modem calls, use the incoming called-number dial peer configuration command. To return to the default value, use the no form of this command. incoming called-number string no incoming called-number string Syntax Description string Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number.
incoming called-number Example 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 58 Release 12.
num-exp num-exp To define a complete telephone number for an extension, use the num-exp global configuration command. Use the no form of this command to cancel a configured number expansion. num-exp extension-number expanded-number no num-exp extension-number Syntax Description extension-number Digit(s) defining an extension number to be expanded. expanded-number Digit(s) defining the expanded telephone number or destination pattern. Defaults No number expansion is configured.
num-exp The following example specifies that all five-digit extensions beginning with 5 be expanded to 1408555 . . . . num-exp 5.... 1408555.... Related Commands 60 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. Release 12.
session target session target To configure a network-specific address for a dial peer, use the session target dial-peer configuration command. Use the no form of this command to disable this feature. Cisco MC3810 Voice over IP: session target {ipv4:destination-address | dns:[$s$. | $d$. | $e$. | $u$.] host-name | loopback:rtp | loopback:compressed | loopback:uncompressed} no session target Syntax Description For the Cisco MC3810 Voice over IP: ipv4:destination-address IP address of the dial peer.
session target Command History 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.
session target 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.
show call active voice show call active voice To show the active call table, use the show call active voice privileged EXEC command. show call active voice Syntax Description This command has no arguments or keywords. Command Mode User EXEC and Privileged EXEC Command History Release Modification 11.3(1)T This command was introduced on the Cisco 2600 series and 3600 series. 12.0(3)XG Support for VoFR was added. 12.0(4)T This command was first supported on the Cisco 7200 series. 12.
show call active voice 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 Voi
show call active voice Table 2 Show Call Active Voice Field Descriptions (continued) Field Description 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.
show call history voice show call history voice To display the call history table, use the show call history voice privileged EXEC command. show call history voice [last number | brief] Syntax Description 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.
show call history voice Example The following is sample output from the show call history voice command for a VoFR call using the frf11-trunk session protocol: router# show call history voice last 1 GENERIC: 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 TransmitByte
show call history voice ACOMLevel=20 SessionTarget= The following is sample output from the show call history voice command for a VoIP call: router# show call history voice GENERIC: 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] Rem
show call history voice Table 3 70 Show Call History Voice Field Descriptions 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.
show call history voice Table 3 Show Call History Voice Field Descriptions (continued) Field Description RemoteIPAddress Remote system IP address for the call. RemoteUDPPort Remote system UDP listener port to which voice packets for this call are transmitted. RoundTripDelay Voice packet round trip delay between the local and remote system on the IP backbone for this call. SelectedQoS Selected quality of service for the call.
show num-exp show num-exp To show the number expansions configured, use the show num-exp privileged EXEC command. show num-exp [dialed-number] Syntax Description dialed-number (Optional) Dialed number. Command Mode User EXEC and Privileged EXEC Command History 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.
show num-exp Table 4 Show Num-Exp Voice Field Descriptions Field Description Dest Digit Pattern Index number identifying the destination telephone number digit pattern. Translation Expanded destination telephone number digit pattern. Related Commands Command Description show call active voice Displays the contents of the active call table. show call history voice Displays the call history table. show dial-peer voice Displays configuration information for dial peers.
voice class codec voice class codec To enter voice-class configuration mode and assign an identification tag number for a codec voice class, use the voice class codec global configuration command. Use the no form of this command to delete a codec voice class. voice class codec tag no voice class codec tag Syntax Description tag The unique number you assign to the voice class. The valid range is 1 to 10000. Each tag number must be unique on the router.
voice class codec Related Commands Command Description codec preference Defines the order of preference in which network dial peers select codecs. voice-class codec (dial-peer) Assigns a previously-configured codec selection preference list to a dial peer.
voice-class codec (dial-peer) voice-class codec (dial-peer) To assign a previously-configured codec selection preference list (codec voice class) to a VoIP dial peer, enter the voice-class codec dial-peer configuration command. Enter the no form of this command to remove the codec preference assignment from the dial peer. voice-class codec tag no voice-class codec tag Syntax Description tag The unique number assigned to the voice class. The valid range for this tag is 1 to 10000.
voice-class codec (dial-peer) Related Commands Command Description codec preference Defines the order of preference in which network dial peers select codecs. voice class codec Enters voice-class configuration mode and assigns an identification tag number for a codec voice class. show dial-peer voice Displays the configuration for all dial peers configured on the router.
voice-group voice-group This command was added in Cisco IOS Release 11.3(1)MA on the Cisco MC3810. Beginning with Cisco IOS Release 12.0(7)XK, this command is no longer supported. 78 Release 12.
