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Voice over ATM on Cisco 3600 series routers extends support for Voice over ATM, previously available only on the Cisco MC3810 to the Cisco 3600 series routers. Voice over ATM enables a Cisco 3600 series router to carry voice traffic (for example, telephone calls and faxes) over an ATM network.
Voice over ATM enables a Cisco 3600 to carry voice traffic, (for example, telephone calls and faxes) over an ATM network by using ATM encapsulation AAL5.
Before you can configure your router to use Voice over ATM, you must first:
After you analyze your dial plan and decide how to integrate it into your existing ATM network, you are ready to configure your network devices to support Voice over ATM.
This section describes the following tasks required to configure Voice over ATM on the Cisco 3600 series routers:
This section describes how to configure voice ports to support Voice over ATM. For more information about voice-port enhancements in Cisco IOS Release 12.0(7)XK, see the online document Voice Port Enhancements in Cisco 2600 and 3600 Series Routers and MC3810 Series Concentrators.
This section is divided into the following procedures:
Voice ports provide support for three basic voice signaling formats:
In general, voice-port commands define the characteristics associated with a particular voice-port signaling type. Under most circumstances, the default voice-port command values are adequate to configure FXO and FXS ports to transport voice data. Because of the inherent complexities involved with PBX networks, E&M ports might need specific voice-port values configured, depending on the specifications of the devices in your telephony network.
The Cisco 3600 Series routers provide analog and digital voice ports. The type of signaling associated with these analog voice ports depend on the interface module installed into the device. The Cisco 3600 series router supports either a two-port or four-port voice network module (VNM); VNMs can hold either two or four voice interface cards (VICs).
Each VIC is specific to a particular signaling type; therefore, VICs determine the type of signaling for the voice ports on that particular VNM. This means that even though VNMs can hold multiple VICs, each VIC on a VNM must conform to the same signaling type. For more information about the physical characteristics of VNMs and VICs or how to install them, refer to the installation document, Voice Network Module and Voice Interface Card Configuration Note, that came with your VNM.
Under most circumstances the default voice-port values are adequate for both FXO and FXS voice ports. To configure FXO and FXS voice ports , use the following commands beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port
For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group | Identify the voice port you want to configure and enter voice-port configuration mode. | ||
| Router(config-voiceport)# dial-type {pulse | dtmf} | (For FXO ports only) Select the appropriate dial type for out-dialing. The default is dtfm. | ||
| Router(config-voiceport)# signal {loop-start | ground-start} | Configure the signaling type for analog FXO and FXS voice ports. The default is loop-start. | ||
| Router(config-voiceport)# compand-type {u-law | a-law} | |||
| Router(config-voiceport)# cptone locale | Configure the appropriate call progress tone for the local region. The default for this command is us. | ||
| Router(config-voiceport)# ring number number | (For FXO ports only) Configure the number of rings detected before a connection is closed on the FXO port. | ||
| Router(config-voiceport)# ring frequency number | (For FXS ports only) Specify the local ring frequency for the FXS voice port. | ||
| Router(config-voiceport)# music-threshold number | (Optional) Specify the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30. | ||
| Router(config-voiceport)# no battery reversal | (Optional) Disable battery polarity reversal on an FXO or FXS port. This command is enabled by default. For more information, see the Cisco IOS Release 12.0(7)XK document Voice Port Enhancements in Cisco 2600 and 3600 Series Routers and MC3810 Series Concentrators. | ||
| Router(config-voiceport)# no disconnect-ack | (Optional) Configure an FXS voice port to not return an acknowledgment upon receipt of a disconnect signal. This command is enabled by default. | ||
| Router(config-voiceport)# connection {plar | tie-line | plar-opx} string | (Optional) Configure the voice-port connection mode type and the destination telephone number. The plar value is used for Private Line Auto Ringdown (PLAR) connections. The tie-line value on the Cisco MC3810 is used for a tie-line connection to a PBX. The plar-opx value is used for PLAR Off-Premises eXtension, to allow the local voice port to provide a local response before the remote voice port receives an answer. Note If you will be configuring Cisco-trunk permanent connections, do not configure this command now. See the "Configuring Cisco-Trunk Permanent Calls on a Cisco 3600 Series Router" section. | ||
| Router(config-voiceport)# description string | |||
| Router(config-voiceport)# voice vad-time milliseconds | (Optional) Change the minimum silence detection time for voice activity detection (VAD). The range is 250 to 65536. The default is 250. | ||
| Router(config-voiceport)# comfort-noise | (Optional) Specify that background noise will be generated if you have VAD activated. |
You can check the validity of your voice-port configuration by performing the following tasks:
Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain and output attenuation for FXO or FXS voice ports. Collectively, these commands are referred to as voice-port tuning commands.
To fine-tune FXO or FXS voice ports , perform the following steps beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port
For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group | Identify the voice port you want to configure and enter voice-port configuration mode. | ||
| Router(config-voiceport)# input gain value | Specify (in decibels) the amount of gain to be inserted at the receiver side of the interface. Acceptable values are from -6 to 14. | ||
| Router(config-voiceport)# output attenuation value | Specify (in decibels) the amount of attenuation at the transmit side of the interface. Acceptable values are from 0 to 14. | ||
| Router(config-voiceport)# echo-cancel enable | Enable echo-cancellation of voice that is sent out the interface and received back on the same interface. | ||
| Router(config-voiceport)# echo-cancel coverage value | Adjust the size (in milliseconds) of the echo-cancel. Acceptable values are 16, 24, and 32. | ||
| Router(config-voiceport)# timeouts initial seconds | Configure the initial timeout value. The initial timeout value specifies the number of seconds the system waits for the caller to input the first digit of the dialed digits. The default is 10 seconds. | ||
| Router(config-voiceport)# timeouts interdigit seconds
| |||
| Router(config-voiceport)# timeouts ringing {seconds | infinity} | Configure the timeout value for ringing. The range is 5 to 60000. The default is 180. | ||
| Router(config-voiceport)# timeouts wait-release {seconds | infinity} | |||
| Router(config-voiceport)# timing digit milliseconds
| If the voice-port dial type is DTMF, configure the DTMF digit signal duration. The range of the DTMF digit signal duration is from 50 to 100 milliseconds. The default is 100. | ||
| Router(config-voiceport)# timing guardout milliseconds | (FXO ports only) Specify the guard-out duration of the FXO voice port. The range is 300 to 3000. The default is 2000. | ||
| Router(config-voiceport)# timing percentbreak percent | (FXO ports only) Specify the percentage of the break period for dialing pulses for a voice port. Valid entries are numbers 20 to 80. The default is 50. | ||
| Router(config-voiceport)# timing inter-digit milliseconds | If the voice-port dial type is DTMF, configure the DTMF inter-digit signal duration. The range of the DTMF inter-digit signal duration is from 50 to 500 milliseconds. The default is 100. | ||
| Router(config-voiceport)# timing pulse-digit milliseconds
| If the voice-port dial type is pulse, configure the pulse digit signal duration. The range of the pulse digit signal duration is from10 to 20 milliseconds. The default is 20. | ||
| Router(config-voiceport)# timing pulse-inter-digit milliseconds
| If the voice-port dial type is pulse, configure the pulse inter-digit signal duration. The range of the pulse inter-digit signal duration is from 100 to 1000 milliseconds. The default is 500. | ||
| Router(config-voiceport)# impedance {600r | 600c | 900r | 900c}
| (For FXO ports only) Configure the impedance. The default is 600 ohms real. | ||
| Router(config-voiceport)# ring cadence {[pattern01 | pattern02 | pattern03 | pattern04 | pattern05 | pattern06 | pattern07 | pattern08 | pattern09 | pattern10 | pattern11 | pattern12] [define pulse interval]} | (For FXS ports only) Specify the ring cadence for the FXS voice port. |
Unlike FXO and FXS voice ports, the default E&M voice-port parameters most likely will not be sufficient to enable voice data transmission over your network.
To configure E&M voice ports , use the following commands beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port
For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group | Identify the voice port you want to configure and enter voice-port configuration mode. | ||
| Router(config-voiceport)# dial-type {dtmf | pulse} | |||
| Router(config-voiceport)# operation {2-wire | 4-wire} | |||
| Router(config-voiceport)# type {1 | 2 | 3 | 5} | Select the appropriate E&M interface type. Type 1 indicates the following lead configuration:
Type 2 indicates the following lead configuration:
Type 3 indicates the following lead configuration:
Type 5 indicates the following lead configuration:
| ||
| Router(config-voiceport)# signal {wink-start | immediate | delay-dial} | Configure the signaling type for E&M voice ports. The default is wink-start. | ||
| Router(config-voiceport)# impedance {600c | 600r | 900c | complex1 | complex2} | Specify a terminating impedance. This value must match the specifications from the telephony system to which this voice port is connected. | ||
| Router(config-voiceport)# connection {plar | tie-line | plar-opx} string | (Optional) Configure the voice-port connection mode type and the destination telephone number. The plar value is used for Private Line Auto Ringdown (PLAR) connections. The tie-line value is used for a tie-line connection to a PBX. The plar-opx value is used for PLAR Off-Premises eXtension, to allow the local voice port to provide a local response before the remote voice port receives an answer. Note If you will be configuring Cisco-trunk permanent connections, do not configure this command now. See the "Configuring Cisco-Trunk Permanent Calls on a Cisco 3600 Series Router" section. | ||
| Router(config-voiceport)# compand-type {u-law | a-law} | Configure the companding standard used to convert between analog and digital signals in PCM systems. This command applies to digital voice-posts only. | ||
| Router(config-voiceport)# cptone locale | Configure the appropriate call progress tone for the local region. The default for this command is us. | ||
| Router(config-voiceport)# music-threshold number | (Optional) Specify the threshold (in decibels) for on-hold music. Valid entries are from -70 to -30. | ||
| Router(config-voiceport)# no auto-cut-through | (Optional) Disables call completion when a PBX does not provide an M-lead response. This command is enabled by default. | ||
| Router(config-voiceport)# description string | (Optional) Attach descriptive text about this voice port connection. |
You can check the validity of your voice-port configuration by performing the following tasks:
Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain and output attenuation for E&M voice ports. Collectively, these commands are referred to as voice-port tuning commands.
To fine-tune E&M voice ports , use the following commands beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port
For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group | Identify the voice port you want to configure and enter voice-port configuration mode. | ||
| Router(config-voiceport)# input gain value | |||
| Router(config-voiceport)# output attenuation value | |||
| Router(config-voiceport)# echo-cancel enable | Enable echo-cancellation of voice that is sent out the interface and received back on the same interface. | ||
| Router(config-voiceport)# echo-cancel coverage value | Adjust the size (in milliseconds) of the echo-cancel. Acceptable values are 16, 24, and 32. | ||
| Router(config-voiceport)# non-linear | |||
| Router(config-voiceport)# timeouts initial seconds | |||
| Router(config-voiceport)# timeouts interdigit seconds | |||
| Router(config-voiceport)# timeouts wait-release {value | infinity} | Configure the timeout value for releasing voice ports. This command limits the duration that a voice port stays in the call failure state while the Cisco MC3810 sends a busy tone, reorder tone or out-of-service tone to the port. | ||
| Router(config-voiceport)# timing digit milliseconds | |||
| Router(config-voiceport)# timing inter-digit milliseconds
| If the voice-port dial type is DTMF, configure the DTMF inter-digit signal duration. The range of the DTMF inter-digit signal duration is from 50 to 500 milliseconds. The default is 100. | ||
| Router(config-voiceport)# timing percentbreak percent | Specify the percentage of the break period for dialing pulses for a voice port. Valid entries are numbers 20 to 80. The default is 50. | ||
| Router(config-voiceport)# timing pulse-digit milliseconds
| If the voice-port dial type is pulse, configure the pulse digit signal duration. The range of the pulse digit signal duration is from10 to 20 milliseconds. The default is 20. | ||
| Router(config-voiceport)# timing pulse-inter-digit milliseconds
| If the voice-port dial type is pulse, configure the pulse inter-digit signal duration. The range of the pulse inter-digit signal duration is from 100 to 1000 milliseconds. The default is 500. | ||
| Router(config-voiceport)# timing wink-duration milliseconds | |||
| Router(config-voiceport)# timing wink-wait milliseconds | |||
| Router(config-voiceport)# timing clear-wait milliseconds | |||
| Router(config-voiceport)# timing delay-duration milliseconds | |||
| Router(config-voiceport)# timing delay-start milliseconds | |||
| Router(config-voiceport)# timing percentbreak percent | Configure the timing percent-break value. This value sets the percentage of the break period for a dialing pulse. The default is 50 percent. |
After you have configured the voice port, you need to activate the voice port to bring it online. In fact it is a good idea to cycle the port---meaning to shut the port down and then bring it online again.
To activate a voice port, use the following command in voice-port configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router(config-voiceport)# no shutdown | Activate the voice port. |
To cycle a voice port, use the following commands in voice-port configuration mode
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router(config-voiceport)# shutdown | Deactivate the voice port. | ||
| Router(config)# voice-port slot/subunit/port | Enter voice-port configuration mode. | ||
| Router(config-voiceport)# no shutdown | Activate the voice port. |
This section describes the preliminary ATM configuration tasks necessary to support Voice over ATM.
To configure the Cisco 3600 series to support Voice over ATM, use the following commands beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router(config)# interface ATMslot/number | Enter ATM interface configuration mode. | ||
| Router(config-if)# pvc [name] vpi/vci | Create an ATM PVC for voice traffic and enter virtual circuit configuration mode. | ||
| Router(config-if-atm-pvc)# encapsulation aal5mux voice | Set the encapsulation of the PVC to support voice traffic. Note To configure a PVC to support data traffic, use aal5snap encapsulation. | ||
| Router(config-if-atm-pvc)# vbr-rt peak-rate average-rate [burst] | Configure the peak rate, average rate, and the burst cell size to perform traffic shaping between voice and data PVCs. By using the vbr-rt command , you can configure the variable bit-rate for real-time networks, such as for voice networks. Traffic shaping is necessary, so that the carrier does not discard the incoming calls from the router. 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:
| ||
| Router(config-if-atm-pvc)# exit | Exit ATM virtual circuit configuration mode. The only commands in ATM virtual circuit configuration mode used for ATM voice PVCs are encapsulation aal5mux voice, vbr-rt, and ilmi. | ||
| Router(config-if)# pvc [name] vpi/vci | Create an ATM PVC for data traffic and enter virtual circuit configuration mode. | ||
| Router(config-if-atm-pvc)# encapsulation aal5snap | Set the encapsulation of the PVC to support ATM data traffic. In ATM PVC configuration mode, configure either the ubr, ubr+, or the vbr-nrt traffic shaping commands for the data PVC as appropriate. | ||
| Router(config-if-atm-pvc)# exit | Exit ATM virtual circuit configuration mode. Repeat Steps 6 and 7 for each data PVC configured. | ||
| Router(config-if)# exit | Exit interface configuration mode. | ||
| Router(config)# exit | Exit configuration mode. | ||
| Router# show atm vc | Verify the ATM PVC configuration. |
 | TimeSaver If possible, you might want to configure the ATM dial peers in a back-to-back configuration before separating them across the ATM network. Using a back-to-back configuration, you can test your Voice over ATM and dial-peer configuration to see if you can successfully make a voice connection. Then, when you place both peers on the network, if you cannot make a voice connection, you can isolate the cause as a network problem. |
There is specific information relative to each dial peer that you must identify before you can configure Voice over ATM. One way to do this is to create a peer configuration table.
Figure 1 shows a diagram of a small voice network in which Router 1, with ATM virtual circuit 20, connects a small sales branch office to the main office through Router 2. There are only two devices in the sales branch office that need to be established as dial peers: a basic telephone and a fax machine. Router 2, with an ATM virtual circuit of 40, is the primary gateway to the main office; as such, it needs to be connected to the company's PBX. There are three basic telephones connected to the PBX that need to be established as dial peers in the main office.
Table 1 shows the peer configuration table for the example illustrated in Figure 1.
| Dial Peer | Extension | Prefix | Dest-Pattern | Type | Voice Port | Session Target |
|---|---|---|---|---|---|---|
| Router 1 |
|
|
|
|
|
|
1 | 61111 |
| +13101161111 | POTS | 2/0/0 |
|
2 | 62222 |
| +13101162222 | POTS | 2/0/1 |
|
10 |
|
| +1310117.... | VoATM |
| S2 20 |
| Router 2 |
|
|
|
|
|
|
11 |
|
| +1310116.... | VoATM |
| S2 40 |
3 | 73333 | 7 | +1310117.... | POTS | 2/1/0 |
|
4 | 74444 | 7 | +1310117.... | POTS | 2/1/0 |
|
5 | 75555 | 7 | +1310117.... | POTS | 2/1/0 |
|
The dial plan shown in Table 1 lists a simple dial-peer configuration table with no special configuration for how you forward or playout excess digits. For more information on other options for designing your dial plan and configuring your dial peers to connect with PBXs, see the "Advanced Dial Peer Configuration" section in the "Configuring Voice over Frame Relay" chapter in the 12.0 IOS Voice, Video, and Home Applications Configuration Guide. The concepts described in that section also apply to Voice over ATM.
There are two different kinds of dial peers:
POTS dial peers associate a telephone number with a particular voice port, so that incoming calls for that telephone number can be received. Voice over ATM dial peers point to specific voice-network devices (by associating destination telephone numbers with a specific ATM virtual circuit), so that outgoing calls can be placed. Both POTS and Voice over ATM dial peers are required if you want to both send and receive calls by using Voice over ATM.
Establishing two-way communication by using Voice over ATM requires establishing a specific voice connection between two defined endpoints. As shown in Figure 2, for outgoing calls (from the perspective of the POTS dial peer 1), the POTS dial peer establishes the source (the originating telephone number and voice port) of the call. The Voice over ATM dial peer establishes the destination by associating the destination phone number with a specific ATM virtual circuit.
In the example, the destination pattern 14085554000 string maps to a U.S. phone number 555-4000, with the digit 1 plus the area code (408) preceding the number. When configuring the destination pattern, set the dial string to match the local dial conventions.
To complete the two-way communications loop, configure Voice over ATM dial peer 2 as shown in Figure 3.
The only exception is when both POTS dial peers are connected to the same router, as shown in Figure 4. In this circumstance, because both dial peers share the same destination IP address, you do not need to configure a Voice over ATM dial peer.
When configuring dial peers, you need to understand the relationship between the destination pattern and the session target. The destination pattern represents the pattern for the device at the voice connection endpoint, such as a telephone or a PBX. The session target represents the serial port on the peer router at the other end of the ATM connection. Figure 5 and Figure 6 show the relationship between the destination pattern and the session target, as seen from the perspective of both routers in a Voice over ATM configuration.
To configure POTS peers, use the following commands beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router(config)# dial-peer voice tag pots |
The tag value identifies the dial peer and must be unique on the router. Do not duplicate a specific tag number. | ||
| Router(config-dial-peer)# destination-pattern string | Configure the dial peer's destination pattern. The string is a series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are the digits 0 through 9 and the letters A through D. You can enter the following special characters in the string:
| ||
| Router(config-dial-peer)# port slot/port/subunit | |||
| Router(config-dial-peer)# preference value | |||
|
or
or
| (Optional) If using the forward-digits feature, configure the digit-forwarding method. The range for the number of digits forwarded (num-digit) is 0 to 32. Refer to the "Command Reference" section for an explanation of the command options. In the default condition, dialed digits not matching the destination pattern are forwarded. Note The no state is not the default state. | ||
| Router(config-dial-peer)# prefix string |
To configure additional POTS dial peers, exit dial-peer configuration mode by entering exit and repeat the previous steps.
To configure Voice over ATM dial peers, use the following commands beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router(config)# dial-peer voice tag voatm | Define a Voice over ATM dial peer for Voice over ATM and enter dial-peer configuration mode. The tag value identifies the dial peer and must be unique on the router. Do not duplicate a specific tag number. | ||
| Router(config-dial-peer)# destination-pattern string | |||
| Router(config-dial-peer)# session target ATMx/y pvc [name] | [vpi/]vci]] | Configure the ATM session target for the dial peer. On the Cisco 3600, if you specify a vpi/vci combination, the valid values depend on the network module installed. 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. | ||
| Router(config-dial-peer)# preference value | (Optional) Configure a preference for the Voice over ATM dial peer. The value is a number from 0 to 10 where the lower the number, the higher the preference. | ||
| Router(config-dial-peer)# codec type [bytes bytes] | Specify the voice coder rate of speech and payload size for the dial peer. The default dial peer codec is g729r8. Specifying the payload size by entering the bytes value is optional. Each codec type defaults to a different payload size if you do not specify a value. To obtain a list of the default payload sizes, enter the codec command and the bytes option followed by a question mark (?). | ||
| Router(config-dial-peer)# dtmf-relay | (Optional) If the codec type is a low bit-rate codec such as g729 or g723, specify support for DTMF relay to improve end-to-end transport of DTMF tones. DTMF tones do not always propagate reliably with low bit-rate codecs. DTMF relay is disabled by default. | ||
| Router(config-dial-peer)# signal-type | (Optional) Define the flavor of the ABCD signaling packets that are generated by the voice port and sent to the data network. The signal type must be configured to the same setting at both ends of the permanent voice call. Enter cas to support CAS. Enter cept to support the European CEPT standard (related to MEL CAS). Enter ext-signal to indicate that ABCD signaling packets should not be sent, for configurations where the line signaling information is carried externally to the voice port. Enter transparent (for digital T1/E1 interfaces) to read the ABCD signaling bits directly from the T1/E1 interface without interpretation and to pass them transparently to the data network (this is also known as transparent FRF.11 signaling). | ||
| Router(config-dial-peer)# no vad | (Optional) Disable voice activity detection (VAD) on the dial peer. This command is enabled by default. | ||
| Router(config-dial-peer)# sequence-numbers | (Optional) Enable the voice sequence number if required for your configuration. This command is disabled by default. | ||
| Router(config-dial-peer)# preference value | (Optional) Configure a preference for the VoFR dial peer. The value is a number from 0 to 10 where the lower the number, the higher the preference in hunt groups. | ||
| Router(config-dial-peer)# session protocol cisco-switched | (Optional) Configure the session protocol to support Cisco-trunk permanent (private line) trunk calls. The cisco-switched option is the default setting, and entering this command is not required. If you do not want the dial peer to support Cisco-trunk permanent (private line) trunk calls, enter no session protocol cisco-switched. | ||
| Router(config-dial-peer)# fax rate {2400 | 4800 | 7200 | 9600 | 14400 | disable | voice} | (Optional) Configure the transmission speed (in bps) at which a fax will be sent to the dial peer. The default is voice, which specifies the highest possible transmission speed allowed by the voice rate. |
To configure additional Voice over ATM dial peers, exit dial-peer configuration mode by entering exit, and repeat the previous steps.
After you have configured dial peers, you can configure how the router performs dial peer hunting functions. To configure the dial peer hunting behavior on the router, perform the following steps beginning in configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router(config)# dial-peer hunt hunt-order-number | Specify the hunt selection order for dial peers. | ||
| Router(config)# dial-peer terminator character | (Optional) Designate a special character to be used as a terminator for variable length dialed numbers. |
If using dial peer hunting, there may be situations when you want to disable dial-peer hunting on a specific dial peer. To disable dial-peer hunting on a dial peer, use the following commands beginning in global configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| Router(config)# dial-peer voice tag {pots | voatm} | Enter dial-peer configuration mode for the specified dial peer. | ||
| Router(config-dial-peer)# huntstop | Disable dial-peer hunting on the dial peer. Once you enter this command, no further hunting is allowed if a call fails on the specified dial peer. |
To reenable dial-peer hunting on a dial peer, enter the no huntstop command.
To configure Cisco-trunk permanent calls on a Cisco 3600 series router for Voice over ATM, use the following commands from interface configuration mode:
| Step | Command | Purpose | ||
|---|---|---|---|---|
| For Cisco 3600 series analog voice ports: Router(config)# voice-port slot/subunit/port
For Cisco 3600 series digital voice ports: Router(config)# voice-port slot/port:ds0-group | Identify the voice port you want to configure and enter voice-port configuration mode. | ||
| Router(config-voiceport)# connection trunk destination-string [answer-mode] | Configure the trunk connection and specify the telephone number in the destination-string. When configuring Cisco-trunk permanent calls, one side must be the call initiator (master) and the other side is normally the call answerer (slave). By default, the voice port operates in master mode. Enter the answer-mode keyword to specify that the voice port operates in slave mode. | ||
| Router(config-voiceport)# shutdown | Shut down the voice port. | ||
| Router(config-voiceport)# no shutdown | Reactivate the voice port to enable the trunk connection to take effect. |
Verify that the voice connection is working by doing the following:
You can check the validity of your dial-peer and voice-port configuration by performing the following tasks:
If you are having trouble connecting a call and you suspect the problem is associated with the dial-peer configuration, you can try to resolve the problem by performing the following tasks:
The following is a sample configuration for Voice over ATM on the Cisco 3600 series:
version 12.0 service timestamps debug uptime service timestamps log uptime no service password-encryption ! hostname c3640_1 ! ! ! no ip subnet-zero no ip routing ip wccp version 2 ip host keyer-ultra 171.71.20.62 ! ! ! ! dial-control-mib max-size 500 ! process-max-time 200 ! interface Ethernet0/0 ip address 172.28.129.54 255.255.255.192 ip helper-address 171.71.20.62 no ip directed-broadcast no ip route-cache no ip mroute-cache ! interface Serial0/0 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache no fair-queue ! interface Ethernet0/1 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache shutdown ! interface ATM1/0 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache no atm ilmi-keepalive pvc 1/100 vbr-rt 1000 500 encapsulation aal5mux voice ! no scrambling-payload impedance 120-ohm no fair-queue ! interface ATM1/1 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache no atm ilmi-keepalive pvc 2/100 vbr-rt 1000 500 encapsulation aal5mux voice ! no scrambling-payload impedance 120-ohm no fair-queue ! interface ATM1/1.1 point-to-point no ip directed-broadcast no ip route-cache no ip mroute-cache pvc 3/200 vbr-rt 64 64 4 encapsulation aal5mux voice ! ! interface ATM1/2 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache shutdown no atm ilmi-keepalive no scrambling-payload impedance 120-ohm no fair-queue ! interface ATM1/3 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache shutdown no atm ilmi-keepalive no scrambling-payload impedance 120-ohm no fair-queue ! interface ATM1/4 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache shutdown no atm ilmi-keepalive no scrambling-payload impedance 120-ohm no fair-queue ! interface ATM1/5 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache shutdown no atm ilmi-keepalive no scrambling-payload impedance 120-ohm no fair-queue ! interface ATM1/6 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache shutdown no atm ilmi-keepalive no scrambling-payload impedance 120-ohm no fair-queue ! interface ATM1/7 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache shutdown no atm ilmi-keepalive no scrambling-payload impedance 120-ohm no fair-queue ! interface ATM3/0 no ip address no ip directed-broadcast no ip route-cache no ip mroute-cache map-group atm1 atm clock INTERNAL pvc 2/200 encapsulation aal5snap no atm auto-configuration no atm ilmi-keepalive no atm address-registration no atm ilmi-enable pvc voice 1/100 vbr-rt 5000 2500 encapsulation aal5mux voice ! ! ip default-gateway 172.28.129.1 ip classless ip route 171.70.20.62 255.255.255.255 172.28.129.1 no ip http server ! ! map-list atm1 ip 4.4.4.2 atm-vc 2 broadcast ! map-class frame-relay fr1 ! map-class frame-relay voice no frame-relay adaptive-shaping frame-relay cir 128000 frame-relay bc 128000 snmp-server engineID local 00000009020000107BC778C0 snmp-server community public RO snmp-server community SNMPv2c view v2default RO snmp-server community v2 view v1default RO snmp-server community config view v1default RO snmp-server community voice view v1default RO snmp-server packetsize 4096 snmp-server enable traps snmp snmp-server enable traps casa snmp-server enable traps config snmp-server enable traps voice poor-qov snmp-server host 171.71.128.229 version 2c SNMPv2c config voice snmp snmp-server host 171.71.128.242 version 2c public config voice snmp snmp-server host 171.71.129.16 version 2c public tty frame-relay isdn hsrp config entity envmon bgp rsvp rtr syslog stun sdllc dspu rsrb dlsw sdlc snmp snmp-server host 171.71.129.164 version 2c public config voice snmp ! line con 0 exec-timeout 0 0 transport input none line aux 0 line vty 0 4 session-timeout 10 password apple login ! voice-port 2/0/0 input gain 5 output attenuation 5 ! voice-port 2/0/1 input gain 5 output attenuation 5 ! voice-port 2/1/0 input gain 5 output attenuation 5 ! voice-port 2/1/1 input gain 5 output attenuation 5 ! dial-peer voice 2 pots destination-pattern 4001 ! dial-peer voice 8000 pots destination-pattern 84000 ! dial-peer voice 9000 pots destination-pattern 94000 ! dial-peer voice 9001 pots destination-pattern 94001 ! dial-peer voice 348 voatm destination-pattern 348.... signal-type ext-signal session target ATM3/0 pvc 1/100 ! dial-peer voice 338 voatm destination-pattern 338.... signal-type ext-signal session target ATM1/0 pvc 1/100 ! dial-peer voice 2222 voatm preference 1 session target ATM1/0 pvc 1/100 ! dial-peer voice 9500 voatm destination-pattern 95... session target ATM3/0 pvc 1/100 ! dial-peer voice 8400 pots destination-pattern 84000 ! dial-peer voice 50000 voatm destination-pattern 5264000 session target ATM3/0 pvc 1/100 ! dial-peer voice 10000 pots destination-pattern 5254000 port 2/0/0 ! dial-peer voice 10001 pots destination-pattern 4000789 port 2/1/0 ! num-exp 1 1234 num-exp 2 2234 num-exp 12 34567890 num-exp 55 66666 end
This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.0 command reference publications.
New and modified debug commands can be found in the "Debug Commands" section.
The following new and modified commands are described in this section (modified commands are marked by an asterisk):
In Cisco IOS Release 12.0(1)T or later releases, you can search and filter the output for show and more commands. This functionality is useful when you need to sort through large amounts of output, or if you want to exclude output that you do not need to see.
To use this functionality, enter a show or more command followed by the "pipe" character (|), one of the keywords begin, include, or exclude, and an expression that you want to search or filter on:
command | {begin | include | exclude} regular-expression
See the following example of the show atm vc command where you want the command output to begin with the first line where the expression "PeakRate" appears:
show atm vc | begin PeakRate
For more information on the search and filter functionality, refer to the Cisco IOS Release 12.0(1)T feature module titled CLI String Search.
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" weight 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 (for example, in certain European countries) where telephone numbers can vary in length. When a dialed-number string has been identified as a variable length dialed-number, the system does not place a call until the configured value for the timeouts interdigits command has expired, or until the caller dials the terminating character. Use the dial-peer terminator global configuration command to change the terminating character.
The following example specifies "9" as the terminating character for variable-length dialed numbers:
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. |
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
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 want to restore the default.
For QSIG ISDN connections, entering forward-digits all implies that all of the digits of the called party number are sent to the ISDN connection. When you enter forward-digits num-digit and enter a number from 1 to 32, the number of digits specified (right justified) of the called part number are sent to the ISDN connection.
The following example forwards all of the digits in the destination pattern of a POTS dial peer:
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. |
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 voatm
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 voatm
router(config-dial-peer)# no huntstop
| Command | Description |
dial-peer voice | Enters dial-peer configuration mode and specifies the method of voice-related encapsulation. |
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.
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 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. |
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 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. |
signal-type | Sets the signaling type to be used when connecting 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. |
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 |
This section provides information on new and modified VoATM debug commands for the Cisco 3600 series.
All other debug commands used with Voice over ATM are documented in the Cisco IOS Release 12.0 Command References.
The following new and modified commands are described in this section:
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. |
AAL---ATM Adaptation Layer.
Call leg---A logical connection between the router and either a telephony endpoint over a bearer channel or another endpoint using a session protocol.
CAS---Channel associated signaling.
Codec---Coder-decoder. Device that typically uses pulse code modulation to transform analog signals into a digital bit stream and digital signals back into analog signals. In Voice over ATM, it specifies the voice coder rate of speech for a dial peer.
Dial peer---An addressable call endpoint. In Voice over ATM, there are two kinds of dial peers: POTS and VoATM.
DS0---A 64-K B-channel on an E1 or T1 WAN interface.
DTMF---Dual tone multifrequency. Use of two simultaneous voice-band tones for dial (such as touch tone).
E&M---E&M stands for recEive and transMit (or Ear and Mouth). E&M is a trunking arrangement generally used for two-way switch-to-switch or switch-to-network connections. Cisco's E&M interface is an RJ-48 connector that allows connections to PBX trunk lines (tie lines).
FXO---Foreign Exchange Office. An FXO interface connects to the PSTN's central office and is the interface offered on a standard telephone. Cisco's FXO interface is an RJ-11 connector that allows an analog connection to be directed at the PSTN's central office. This interface is of value for off-premise extension applications.
FXS---Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies ring, voltage, and dial tone. Cisco's FXS interface is an RJ-11 connector that allows connections to basic telephone service equipment, keysets, and PBXs.
PBX---Private Branch Exchange. Privately owned central switching office.
PLAR---Private Line Auto Ringdown. This type of service results in a call attempt to some particular remote endpoint when the local extension is taken off-key.
POTS---Plain Old Telephone Service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the public switched telephone network.
POTS dial peer---Dial peer connected via a traditional telephony network. POTS peers point to a particular voice port on a voice network device.
PSTN---Public Switched Telephone Network. PSTN refers to the local telephone company.
PVC---Permanent virtual circuit.
TDM---Time division multiplexing.
Trunk---Service that allows quasi-transparent connections between two PBXs, a PBX and a local extension, or some other combination of telephony interfaces to be permanently conferenced together by the session application and signaling passed transparently through the IP network.
VBR---Variable Bit Rate.
VoATM dial peer---Dial peer connected by an ATM network. VoATM peers point to specific VoATM devices.
Posted: Thu Mar 30 16:03:41 PST 2000
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