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Table Of Contents
Voice over IP for the Cisco AS5800
Related Features and Technologies
Supported Standards, MIBs, and RFCs
Configuring IP Networks for Real-Time Voice Traffic
Configuring Custom Queuing and IP RTP Reserve
Verifying Voice Port Configuration
Inbound versus Outbound Dial Peers
Outbound Dialing on POTS Peers
Direct Inward Dial for POTS Peers
Distinguishing Voice and Modem Calls on the Cisco AS5800
Verifying Dial Peer Configuration
Configuring the Cisco AS5800 as an H.323 Gateway
Verifying Gateway Interface Configuration
Configuring the Cisco AS5800 for Interactive Voice Response
Configuring the Cisco 3640 as a Gatekeeper
Configuring the Cisco 2600 as a Gateway
Configuring the Cisco AS5800 as a Gateway
Voice over IP for the Cisco AS5800
The Voice over IP for the Cisco AS5800 feature adds Voice over IP carrier-class gateway functionality to the Cisco AS5800 platform. This document contains the following sections:
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Supported Standards, MIBs, and RFCs
Feature Overview
Voice over IP (VoIP) enables a Cisco AS5800 universal access server to provide voice and fax traffic, such as telephone calls and faxes, over an IP network. There are basically two different environments in which VoIP can be deployed: enterprise and service provider. Different strategies have been developed for deploying VoIP in both of these environments. The Cisco AS5800 universal access server can be configured for deployment in either an enterprise or a service provider environment but, because of the extensive capabilities of the Cisco AS5800 universal access server, it is more likely that it will function as a carrier class gateway in a service provider environment. This document, then, describes how to configure the Cisco AS5800 universal access server to act as a carrier class gateway in your VoIP network. To configure the Cisco AS5800 universal access server to perform in an enterprise environment, refer to the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module. The configuration steps for both the Cisco AS5300 access server and the Cisco AS5800 universal access server for an enterprise environment are identical.
Voice over IP in either the service provider or enterprise environment is primarily a software feature; however, to use this feature on the Cisco AS5800, you must install a VoIP feature card (VFC). The VFC uses the Cisco AS5800's T1/E1 and T3 Public Switched Telephone Network (PSTN) interfaces and local-area network (LAN) or wide-area network (WAN) routing capabilities to provide up to a 192 ports or channels (per VFC card) for VoIP packetized voice traffic.
Benefits
Two-Stage-Dial Toll Bypass
With Voice over IP on the Cisco AS5800, you can leverage your network's WAN infrastructure to offer long distance toll bypass services. Toll bypass occurs in two stages. For example, customers can be assigned an account number and a Personal Identification Number (PIN). When a user dials a local number or a 1-800-Internet Telephone Service Provider (ITSP) number, she connects to the local VoIP point of presence. She is then prompted by the Interactive Voice Response (IVR) to input her account and PIN numbers. Following authentication, a second dial tone allows her to enter an E.164 destination telephone number.
The local gatekeeper maps the E.164 destination telephone number to an IP address of a remote-zone gatekeeper, which then selects a gateway to terminate the call. The gateway encodes the call, encapsulates it in Real Time Protocol (RTP) packets and routes it over the WAN to the remote gateway. The remote gateway decodes the call and delivers it to the receiver.
For information about configuring IVR, refer to the Cisco IOS Release 12.0(7)T Configuring Interactive Voice Response for Cisco Access Platforms feature module.
Figure 1 Two-Stage Dial Toll Bypass
PSTN Voice-Traffic and Fax-Traffic Off load
Carriers can leverage their WAN infrastructure to off load voice and fax traffic from their congested PSTN networks by using the Cisco AS5800 as a carrier class voice gateway. In this application, PSTN traffic designated to be off-loaded is forwarded to a tandem switch connected to the Cisco AS5800 gateway. The AS5800 gateway then encapsulates the off-loaded PSTN traffic into RTP streams and routes it across the WAN.
The signaling interface between the PSTN and the Cisco AS5800 can be either Common Channel Signaling (CCS), with SS7 terminated by the VCO-4K service point or Channel Associated Signaling (CAS), configured in Direct Inward Dial (DID) mode. illustrates this application.
Figure 2 VoIP Used as a PSTN Gateway to Off load Voice Traffic and Fax Traffic
Universally Accessible Voice-Mail and Fax-Mail Services
VoIP on the Cisco AS5800 can be used to leverage the technology prefixes feature. Gateways (with voice/fax feature cards) that are connected to the voice-mail and fax-mail servers can be identified by gatekeepers based on a prefix prepended to an E.164 telephone number.
Additional Benefits
VoIP on the Cisco AS5800 can be used to provide the following additional benefits:
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Restrictions
To run Voice over IP on the Cisco AS5800, the AS5800 must have a version of the Cisco IOS software installed that supports DSDWare 3.1.7 (for example, Cisco IOS Release 12.0(4)XL or Cisco IOS Release 12.0(7)T).
Related Features and Technologies
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Related Documents
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Supported Platforms
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Supported Standards, MIBs, and RFCs
Standards
None
MIBs
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For descriptions of supported MIBs and how to use MIBs, see the Cisco MIB web site on CCO at http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml.
RFCs
None
Prerequisites
Before you can configure your Cisco AS5800 to use Voice over IP, you must first:
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For more information about any of the these configuration tasks, refer to the Cisco AS5800 Universal Access Server Software Installation and Configuration Guide, which shipped with your Cisco AS5800 and is available on the document CD-ROM.
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Configuration Tasks
After you have analyzed your dial plan and decided how to integrate it into your existing IP network, you are ready to configure your network devices to support Voice over IP. The actual configuration procedure depends entirely on the topology of your voice network, but, in general, you need to complete the following tasks:
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Configuring IP Networks for Real-Time Voice Traffic
You need to have a well-engineered network end-to-end when running delay-sensitive applications such as VoIP. Fine-tuning your network to adequately support VoIP involves a series of protocols and features geared toward Quality of Service (QoS). It is beyond the scope of this document to explain the specific details relating to wide-scale QoS deployment. Cisco IOS software provides many tools for enabling QoS on your backbone, such as Random Early Detection (RED), Weighted Random Early Detection (WRED), Fancy Queuing (meaning custom, priority, or weighted fair queuing), and IP Precedence. To configure your IP network for real-time voice traffic, you need to take into consideration the entire scope of your network, then select the appropriate QoS tool or tools. In addition, you must use the Cisco IOS ip cef command to ensure that Cisco Express Forwarding (CEF) is enabled.
QoS must be configured throughout your network—not just on the Cisco AS5800 devices running VoIP—to improve voice network performance. Not all QoS techniques are appropriate for all network routers. Edge routers and backbone routers in your network do not necessarily perform the same operations; the QoS tasks they perform might also differ. To configure your IP network for real-time voice traffic, you need to consider the functions of both edge and backbone routers in your network, then select the appropriate QoS tool or tools.
In general, edge routers perform the following QoS functions:
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In general, backbone routers perform the following QoS functions:
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Scalable QoS solutions require cooperative edge and backbone functions.
Configuring Custom Queuing and IP RTP Reserve
Although not required, you can use the custom queuing QoS tool to fine-tune your network for real-time voice traffic. Real-time voice traffic is carried on UDP ports ranging from 16384 to 32767. Custom Queuing and other methods for identifying high priority streams should be configured for these port ranges. For more information about custom queuing, refer to the "Congestion Management" chapter in the Cisco IOS Release 12.0 Quality of Service Configuration Guide. For more information about configuring IP RTP Priority, refer to the Cisco IOS Release 12.0(5)T IP RTP Priority feature module.
Configuring Voice Ports
When an ISDN interface on the Cisco AS5800 is carrying voice data, it is referred to as a voice port.
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Signaling in Voice over IP for the AS5800 is handled by ISDN PRI group configuration. After ISDN PRI is configured for both B and D channels for both ISDN PRI lines, you need to issue the isdn incoming-voice command on the serial interface (acting as the D channel) to ensure a dial tone.
Under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. Because of the inherent complexities involved with PBX networks, you might need specific voice-port values configured, depending on the specifications of the devices in your telephony network. For more information on specific voice-port configuration commands, refer to either the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module or the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference.
To configure basic ISDN parameters for Voice over IP on the Cisco AS5800, perform the following steps:
As mentioned, under most circumstances, the default voice-port command values are adequate to configure voice ports to transport voice data over your existing IP network. If you need to configure specific voice port parameters, perform the following steps beginning in privileged EXEC mode:
Fine-Tuning ISDN Voice Ports
Depending on the specifics of your particular network, you may need to adjust voice parameters involving timing, input gain, and output attenuation for voice ports. Collectively, these commands are referred to as voice-port tuning commands.
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To fine-tune ISDN voice ports, use the following commands beginning in privileged EXEC mode:
For more information on specific voice-port configuration commands or additional voice-port commands, refer to either the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module or the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference..
Verifying Voice Port Configuration
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Troubleshooting Tips
If you are having trouble connecting a call, and you suspect the problem is associated with voice-port configuration, you can try to resolve the problem by performing the following tasks:
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Configuring Dial Peers
The key point to understanding how VoIP functions is to understand dial peers. Each dial peer defines the characteristics associated with a call leg, as shown in and . A call leg is a discrete segment of a call connection that lies between two points in the connection. All of the call legs for a particular connection have the same connection ID.
There are two different kinds of dial peers:
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An end-to-end call comprises four call legs, two from the perspective of the source access server as shown in , and two from the perspective of the destination access server as shown in . A dial peer is associated with each call leg. Dial peers are used to apply attributes to call legs and to identify call origin and destination. Attributes applied to a call leg include QoS, codec, VAD, and fax rate.
Figure 3 Dial Peer Call Legs from the Perspective of the Source Router
Figure 4 Dial Peer Call Legs from the Perspective of the Destination Router
Inbound versus Outbound Dial Peers
Dial peers are used for both inbound and outbound call legs. It is important to remember that these terms are defined from the access server's perspective. An inbound call leg originates outside the access server. An outbound call leg originates from the access server.
For inbound call legs, a dial peer might be associated to the calling number or the port designation. Outbound call legs always have a dial peer associated with them. The destination pattern is used to identify the outbound dial peer. The call is associated with the outbound dial peer at setup time.
POTS peers associate a telephone number with a particular voice port so that incoming calls for that telephone number can be received and outgoing calls can be placed. VoIP peers point to specific devices (by associating destination telephone numbers with a specific IP address) so that incoming calls can be received and outgoing calls can be placed. Both POTS and VoIP peers are needed to establish VoIP connections.
Configuring POTS Peers
POTS peers enable incoming calls to be received by a particular telephony device. To configure a POTS peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its telephone numbers, and associate it with a voice port through which calls will be established. Under most circumstances, the default values for the remaining dial peer configuration commands will be sufficient to establish connections.
To configure a POTS dial peer, use the following commands beginning in global configuration mode:
For additional POTS dial-peer configuration commands, refer to the "Voice-Related Commands" section of the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference, the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module, and the Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module.
Outbound Dialing on POTS Peers
When a router receives a voice call, it selects an outbound dial peer by comparing the called number (the full E.164 telephone number) in the call information with the number configured as the destination pattern for the POTS peer. The router then strips out the left-justified numbers corresponding to the destination pattern matching the called number. If you have configured a prefix, the prefix will be put in front of the remaining numbers, creating a dial string, which the router will then dial. If all numbers in the destination pattern are stripped-out, the user will receive (depending on the attached equipment) a dial tone.
For example, suppose there is a voice call whose E.164 called number is 1 310 767-2222. If you configure a destination-pattern of "1310767" and a prefix of "9," the router will strip out "1310767" from the E.164 telephone number, leaving the extension number of "2222." It will then append the prefix, "9," to the front of the remaining numbers, so that the actual numbers dialed is "9, 2222." The comma in this example means that the router will pause for one second between dialing the "9" and the "2" to allow for a secondary dial tone.
Direct Inward Dial for POTS Peers
Direct inward dial (DID) is used to determine how the called number is treated for incoming POTS call legs. As shown in , incoming means from the perspective of the router. In this case, it is the call leg coming into the access server to be forwarded through to the appropriate destination pattern.
Figure 5 Incoming and Outgoing POTS Call Legs
Unless otherwise configured, when a call arrives on the access server, the server presents a dial tone to the caller and collects digits until it can identify the destination dial peer. After the dial peer is identified, the call is forwarded through the next call leg to the destination.
There are cases where it might be necessary for the server to use the called-number (DNIS) to find a dial peer for the outgoing call leg—for example, if the switch connecting the call to the server has already collected the digits. DID enables the server to match the called-number with a dial peer and then directly place the outbound call. With DID, the server does not present a dial tone to the caller and does not collect digits; it forwards the call directly to the configured destination.
To use DID and incoming called-number, a dial peer must be associated with the incoming call leg. Before doing this, it helps if you understand the logic behind the algorithm used to associate the incoming call leg with the dial peer.
The algorithm used to associate incoming call legs with dial peers uses three inputs (which are derived from signaling and interface information associated with the call) and four defined dial peer elements. The three signaling inputs are:
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The four defined dial peer elements are:
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Using the elements, the algorithm is as follows:
For all peers where call type (VoIP versus POTS) match dial peer type:if the type is matched, associate the called number with the incoming called-numberelse if the type is matched, associate calling-number with answer-addresselse if the type is matched, associate calling-number with destination-patternelse if the type is matched, associate voice port to portThis algorithm shows that if a value is not configured for answer-address, the origin address is used because, in most cases, the origin address and answer-address are the same.
To configure a POTS dial peer for direct inward dial, use the following commands beginning in global configuration mode:
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Distinguishing Voice and Modem Calls on the Cisco AS5800
When the Cisco AS5800 is handling both modem and voice calls, it needs to be able to identify the service type of the call—that is, whether or not the incoming call to the server is a modem or a voice call. When the access server handles only modem calls, the service type identification is handled through modem pools. Modem pools associate calls with modem resources based on the called-number (DNIS). In a mixed environment, where the server receives both modem and voice calls, you need to identify the service type of a call by using the incoming called-number command.
Without this, the server attempts to resolve whether an incoming call is a modem or voice call based on the interface over which the call comes. If the call comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes in over a voice port associated with a dial peer, the call is assumed to be a voice call.
It helps to understand the logic behind the algorithm the system uses to distinguish voice and modem calls. The algorithm is as follows:
If the called-number matches a number from the modem pool, handle the call as a modem callIf the called-number matches a configured dial peer incoming called number, handle the call as a voice callElse handle the call as a modem call by default modem poolIf there is no called-number information configured, call classification is handled as follows:
If the interface matches the interface configured for the modem pool, handle the call as a modem call.If the voice port matches the one configured as the dial peer port, handle the call as a voice callElse handle the call as a modem call by default modem poolTo identify the service type of a call to be voice, use the following commands beginning in global configuration mode:
Configuring VoIP Peers
VoIP peers enable outgoing calls to be made from a particular telephony device. To configure a VoIP peer, you need to uniquely identify the peer (by assigning it a unique tag number), define its destination telephone number and destination IP address. As with POTS peers, under most circumstances, the default values for the remaining dial peer configuration commands will be adequate to establish connections.
To configure a VoIP peer, use the following commands beginning in global configuration mode:
For additional VoIP dial peer configuration options, refer to the "Voice-Related Commands" section of the Cisco IOS Release 12.0 Voice, Video, and Home Applications Command Reference, the Cisco IOS Release 12.0(3)T Voice over IP for the Cisco AS5300 feature module, and the Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module.
Verifying Dial Peer Configuration
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Troubleshooting Tips
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Configuring the Cisco AS5800 as an H.323 Gateway
The Service Provider implementation of Voice over IP uses both gatekeepers and gateways. Because of the extensive capabilities of the Cisco AS5800 universal access server, it is likely that it will function as a carrier class gateway in a Service Provider environment. The final step in configuring the Cisco AS5800 for Voice over IP functionality is to configure one of its interfaces as a gateway interface. You can use either an interface that is connected to the gatekeeper or a loopback interface for the gateway interface. The interface that is connected to the gatekeeper is usually a LAN interface—Fast Ethernet, Ethernet, FDDI, or Token Ring.
To configure a gateway interface, perform the following steps beginning in the global configuration mode:
For more information about configuring gateways and gatekeepers, refer to the Cisco IOS Release 12.0(3)T Service Provider Features for Voice over IP feature module.
Verifying Gateway Interface Configuration
Use the show gateway command to find the current registration information and status of the gateway.
Configuring the Cisco AS5800 for Interactive Voice Response
The Interactive Voice Response (IVR) Service Provider application provides IVR capabilities using Tool Command Language (TCL) scripts. For example, an IVR script is played when a caller receives a voice-prompt instruction to enter a specific type of information, such as a PIN. After playing the voice prompt, the IVR application collects the predetermined number of touch tones (digit collection) and forwards the collected digits to a server for storage and retrieval. Call records can be kept, and a variety of accounting functions performed.
Available IVR Scripts
The following is a description of the available IVR scripts:
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The caller can interrupt the message by entering digits for the account number which will trigger the prompt to enter the PIN number. If authentication fails the third time, the script plays the audio file auth_fail_final.au, then hangs up.
Configuring IVR
To use IVR with scripts, you need to configure the inbound POTS dial peer to support IVR, as well as enable IVR functionality by using the call application global configuration. To configure IVR, use the following commands beginning in the global configuration mode:
For more information about configuring IVR, refer to the Cisco IOS Release 12.0(7)T Configuring Interactive Voice Response for Cisco Access Platforms feature module.
Verifying IVR Configuration
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Configuration Example
The following configuration example shows an abbreviated configuration using a Cisco 2600 router and a CiscoAS5800 universal access server as gateways and a Cisco 3600 router as a gatekeeper. shows the network diagram for this particular scenario.
Figure 6 AS5800 Universal Access Server Acting as a Gateway
Configuring the Cisco 3640 as a Gatekeeper
! Configure the Ethernet interface to be used at the gatekeeper interface.interface Ethernet0/1ip address 172.30.00.00 255.255.255.0no ip directed-broadcastno logging event link-statusno keepalive!! Configure the gatekeeper interface and enable the interface.gatekeeperzone local gk3.gg-dn1 gg-dn1 173.50.00.00zone prefix gk3.gg-dn1 21*gw-type-prefix 9#* gw ipaddr 173.60.0.0 1720gw-type-prefix 6#* gw ipaddr 173.60.0.199 1720no use-proxy gk3.gg-dn1 default inbound-to terminalno shutdown!Configuring the Cisco 2600 as a Gateway
! Configure POTS and VoIP dial peers.dial-peer voice 88 voipdestination-pattern 11111tech-prefix 9#session ras!dial-peer voice 11 potsincoming called-number 11111destination-pattern 6#12345port 1/1/1prefix 12345!! Configure the gateway interface.interface Ethernet0/0ip address 173.60.0.199 255.255.255.0no ip directed-broadcastno ip mroute-cacheno logging event link-statusno keepaliveno cdp enabledh323-gateway voip interfaceh323-gateway voip id gk3.gg-dn1 ipaddr 173.30.0.0 1719h323-gateway voip h323-id gw6@gg-dn1h323-gateway voip tech-prefix 6#!Configuring the Cisco AS5800 as a Gateway
! Configure the T1 controller. (This configuration is for a T3 card.)controller T1 1/0/0:1framing esflinecode b8zspri-group timeslots 1-24!! Configure POTS and VoIP dial peers.dial-peer voice 11111 potsincoming called-number 12345destination-pattern 9#11111direct-inward-dialport 1/0/0:1:Dprefix 11111!dial-peer voice 12345 voipdestination-pattern 12345tech-prefix 6#session target ras!! Enable gateway functionality.gateway!! Enable Cisco Express Forwarding.ip cef!! Configure and enable the gateway interface.interface FastEthernet0/3/0ip address 173.60.0.0.255.255.255.0no ip directed-broadcastno keepalivefull-duplexno cdp enableh323-gateway voip interfaceh323-gateway voip id gk3.gg-dn1 ipaddr 173.30.0.0 1719h323-gateway voip h323-id gw3@gg-dn1h323-gateway voip tech-prefix 9#!! Configure the serial interface.(This configuration is for a T3 serial interface.)interface Serial1/0/0:1:23no ip addressno ip directed-broadcastip mroute-cacheisdn switch-type primary-5essisdn incoming-voice modemno cdp enableCommand Reference
This section documents new or modified commands. All other commands used with this feature are documented in one of the following Cisco IOS documentation:
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New Commands
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Modified Commands
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In Cisco IOS Release 12.0(1)T or later, 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
Following is an example of the show atm vc command in which 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.
codec
To specify the voice coder rate of speech for a dial peer, use the codec dial-peer configuration command. To restore the default voice coder rate of speech value, use the no form of this command.
codec {g711alaw | g711ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 |
g728 | g729abr8 | g729ar8 | g729br8 | g729r8 | gsmfr}
no codec
Syntax Description
Defaults
g729r8.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
For toll quality, use the g711alaw or g711ulaw values. These values provide high-quality voice transmission but use a significant amount of bandwidth. For almost toll quality (and a significant savings in bandwidth), use the g729r8 value.
If codec values for the VoIP peers of a connection do not match, the call will fail.
This command is only applicable to VoIP peers.
Examples
The following example configures a voice coder rate that provides toll quality but uses a relatively high amount of bandwidth:
dial-peer voice 10 voipcodec g711alawRelated Commands
dtmf-relay
Specifies how an H.323 gateway relays DTMF tones between telephony interfaces and an IP network.
dtmf-relay
To specify how an H.323 gateway relays dual tone multifrequency (DTMF) tones between telephony interfaces and an IP network, use the dtmf-relay dial-peer configuration command. To remove all signaling options and transmit the DTMF tones as part of the audio stream, use the no form of this command.
dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal]
no dtmf-relay
Syntax Description
Defaults
No default behavior or values.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
DTMF is the tone generated when you press a digit on a touch-tone phone. This tone is compressed at one end of a call; when the tone is decompressed at the other end, it can become distorted, depending on the codec used. The DTMF relay feature transports DTMF tones generated after call establishment out of band using a standard H.323 out-of-band method and a proprietary RTP-based mechanism.
The gateway sends DTMF tones in the format you specify only if the remote device supports it. If the remote device supports multiple formats, the gateway chooses the format based on the following priority:
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The principal advantage of the dtmf-relay command is that it transmits DTMF tones with greater fidelity than is possible in-band for most low-bandwidth CODECs, such as G.729 and G.723. Without the use of DTMF relay, calls established with low-bandwidth CODECs may have trouble accessing automated DTMF-based systems, such as voice-mail, menu-based ACD systems, and automated banking systems.
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Examples
The following example configures DTMF relay with the cisco-rtp option when sending DTMF tones to dial-peer 103:
5800# configure terminal5800(config)# dial-peer voice 103 voip5800(config-dial-peer)# dtmf-relay cisco-rtp5800(config-dial-peer)# end5800#The next example configures the gateway to send DTMF in-band (the default) when sending DTMF tones to dial-peer 103:
5800# configure terminal5800(config)# dial-peer voice 103 voip5800(config-dial-peer)# no dtmf-relay5800(config-dial-peer)# endRelated Commands
port
To associate a dial peer with a specific voice port, use the port dial peer configuration command. To cancel this association, use the no form of this command.
Cisco 2600/3600 Series Router
port slot/subunit/port
no port
Cisco MC3810
port slot/port
no port
Cisco AS5300 Access Server
port controller number:D
no port
Cisco AS5800 Access Server
port {shelf/slot/port:D} | {shelf/slot/parent:port:D}
no port
Syntax Description
Default
No port is configured.
Command Mode
Dial-peer configuration
Command History
Usage Guidelines
This command is used for calls incoming from a telephony interface to select an incoming dial peer and for calls coming from the VoIP network to match a port with the selected outgoing dial peer.
This command applies only to POTS peers.
Example
The following example associates a Cisco 3600 series router POTS dial peer 10 with voice port 1, which is located on subunit 0, and accessed through port 0:
dial-peer voice 10 potsport 1/0/0The following example associates a Cisco MC3810 POTS dial peer 10 with voice port 0, which is located in slot 1:
dial-peer voice 10 potsport 1/0The following example associates a Cisco AS5300 POTS dial peer 10 with voice port 0:D:
dial-peer voice 10 potsport 0:DThe following example associates a Cisco AS5800 POTS dial peer 10 with voice port 1/0/0:D (T1 card):
dial-peer voice 10 potsport 1/0/0:DThe following example associates a Cisco AS5800 POTS dial peer 10 with voice port 1/0/0:1:D (T3 card):
dial-peer voice 10 potsport 1/0/0:1:Dshow csm
To display the call switching module (CSM) statistics for a particular or all DSP channels or for a specific modem or DSP channel, use the show csm privileged EXEC command.
Cisco AS5300 Access Server
show csm {modem [slot/port | modem-group-number] | voice [slot/dspm/dsp/dsp-channel]}
Cisco AS5800 Universal Access Server
show csm voice [shelf/slot/port]
Syntax Description
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
11.3 NA
This command was introduced.
12.0(3)T
Port-specific values for the Cisco AS5300 were added.
12.0(7)T
Port-specific values for the Cisco AS5800 were added.
Usage Guidelines
This command shows the information related to CSM, which includes the DSP channel, the start time of the call, the end time of the call, and the channel on the controller used by the call.
Use the show csm modem command to display the CSM call statistic information for a specific modem, for a group of modems, or for all modems. If a slot/port argument is specified, then CSM call statistics are displayed for the specified modem. If the modem-group-number argument is specified, the CSM call statistics for all of the modems associated with that modem group are displayed. If no keyword is specified, CSM call statistics for all modems on the AS5300 are displayed.
Use the show csm voice command to display CSM statistics for a particular DSP channel. If the slot/dspm/dsp/dsp-channel or shelf/slot/port argument is specified, the CSM call statistics for calls using the identified DSP channel will be displayed. If no argument is specified, all CSM call statistics for all DSP channels will be displayed.
Examples
The following is sample output from the Cisco AS5300 for the show csm voice command:
Router# show csm voice 2/4/4/0slot 2, dspm 4, dsp 4, dsp channel 0,slot 2, port 56, tone, device_status(0x0002): VDEV_STATUS_ACTIVE_CALL.csm_state(0x0406)=CSM_OC6_CONNECTED, csm_event_proc=0x600E2678, current call thru PRI lineinvalid_event_count=0, wdt_timeout_count=0wdt_timestamp_started is not activatedwait_for_dialing:False, wait_for_bchan:Falsepri_chnl=TDM_PRI_STREAM(s0, u0, c22), tdm_chnl=TDM_DSP_STREAM(s2, c27)dchan_idb_start_index=0, dchan_idb_index=0, call_id=0xA003, bchan_num=22csm_event=CSM_EVENT_ISDN_CONNECTED, cause=0x0000ring_no_answer=0, ic_failure=0, ic_complete=0dial_failure=0, oc_failure=0, oc_complete=3oc_busy=0, oc_no_dial_tone=0, oc_dial_timeout=0remote_link_disc=0, stat_busyout=0oobp_failure=0call_duration_started=00:06:53, call_duration_ended=00:00:00, total_call_duration=00:00:44The calling party phone number = 408The called party phone number = 5271086total_free_rbs_timeslot = 0, total_busy_rbs_timeslot = 0, total_dynamic_busy_rbs_timeslot = 0, total_static_busy_rbs_timeslot = 0,total_sw56_rbs_timeslot = 0, total_sw56_rbs_static_bo_ts = 0,total_free_isdn_channels = 21, total_busy_isdn_channels = 0,total_auto_busy_isdn_channels = 0,min_free_device_threshold = 0The following is sample output from the Cisco AS5800 for the show csm voice command:
5800# show csm voice 1/8/19shelf 1, slot 8, port 19VDEV_INFO:slot 8, port 19vdev_status(0x00000401):VDEV_STATUS_ACTIVE_CALL.VDEV_STATUS_HASLOCK.csm_state(0x00000406)=CSM_OC6_CONNECTED, csm_event_proc=0x60868B8C, currentcall thru PRI lineinvalid_event_count=0, wdt_timeout_count=0watchdog timer is not activatedwait_for_bchan:Falsepri_chnl=(T1 1/0/0:22), vdev_chnl=(s8, c19)start_chan_p=0, chan_p=62436D58, call_id=0x800D, bchan_num=22The calling party phone number =The called party phone number = 7511ring_no_answer=0, ic_failure=0, ic_complete=0dial_failure=0, oc_failure=0, oc_complete=1oc_busy=0, oc_no_dial_tone=0, oc_dial_timeout=0remote_link_disc=0, busyout=0, modem_reset=0call_duration_started=3d16h, call_duration_ended=00:00:00,total_call_duration=00:00:00explains the fields contained in both of these examples.
Related Commands
show voice port
To display configuration information about a specific voice port, use the show voice port privileged EXEC command.
Cisco 2600/3600 Series Router
show voice port slot-number/subunit-number/port
Cisco MC3810
show voice port [slot/port] [summary]
Cisco AS5300 Access Router
show voice port controller number:D
Cisco AS5800 Universal Access Router
show voice port {shelf/slot/port:D} | {shelf/slot/parent:port:D}
Syntax Description
For the Cisco 2600/3600 series:
For the Cisco MC3810:
For the Cisco AS5300 Access Server:
controller number
Specifies the T1 or E1 controller.
:D
Indicates the D channel associated with ISDN PRI.
For the Cisco AS5800 Universal Access Server:
Command Mode
Privileged EXEC
Command History
Usage Guidelines
This command applies to Voice over IP, Voice over Frame Relay, Voice over ATM, and Voice over HDLC.
Use the show voice port privileged EXEC command to display configuration and voice interface card-specific information about a specific port.
Examples
The following is sample output from the show voice port command for an E&M voice port on the Cisco 3600 series:
router# show voice port 1/0/0E&M Slot is 1, Sub-unit is 0, Port is 0Type of VoicePort is E&MOperation State is unknownAdministrative State is unknownThe Interface Down Failure Cause is 0Alias is NULLNoise Regeneration is disabledNon Linear Processing is disabledMusic On Hold Threshold is Set to 0 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is disabledEcho Cancel Coverage is set to 16msConnection Mode is NormalConnection Number isInitial Time Out is set to 0 sInterdigit Time Out is set to 0 sAnalog Info Follows:Region Tone is set for northamericaCurrently processing noneMaintenance Mode Set to None (not in mtc mode)Number of signaling protocol errors are 0Voice card specific Info Follows:Signal Type is wink-startOperation Type is 2-wireImpedance is set to 600r OhmE&M Type is unknownDial Type is dtmfIn Seizure is inactiveOut Seizure is inactiveDigit Duration Timing is set to 0 msInterDigit Duration Timing is set to 0 msPulse Rate Timing is set to 0 pulses/secondInterDigit Pulse Duration Timing is set to 0 msClear Wait Duration Timing is set to 0 msWink Wait Duration Timing is set to 0 msWink Duration Timing is set to 0 msDelay Start Timing is set to 0 msDelay Duration Timing is set to 0 msThe following is sample output from the show voice port command for an FXS voice port on the Cisco 3600 series:
router# show voice port 1/0/0Foreign Exchange Station 1/0/0 Slot is 1, Sub-unit is 0, Port is 0Type of VoicePort is FXSOperation State is DORMANTAdministrative State is UPThe Interface Down Failure Cause is 0Alias is NULLNoise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to 0 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 16msConnection Mode is NormalConnection Number isInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sAnalog Info Follows:Region Tone is set for northamericaCurrently processing noneMaintenance Mode Set to None (not in mtc mode)Number of signaling protocol errors are 0Voice card specific Info Follows:Signal Type is loopStartRing Frequency is 25 HzHook Status is On HookRing Active Status is inactiveRing Ground Status is inactiveTip Ground Status is inactiveDigit Duration Timing is set to 100 msInterDigit Duration Timing is set to 100 msHook Flash Duration Timing is set to 600 msThe following is sample output from the show voice port command for an FXS voice port on the Cisco MC3810:
router# show voice port 1/2Voice port 1/2 Slot is 1, Port is 2Type of VoicePort is FXSOperation State is UPAdministrative State is UPNo Interface Down FailureDescription is not setNoise Regeneration is enabledNon Linear Processing is enabledIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 8 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sCoder Type is g729ar8Companding Type is u-lawVoice Activity Detection is disabledRinging Time Out is 180 sWait Release Time Out is 30 sNominal Playout Delay is 80 millisecondsMaximum Playout Delay is 160 millisecondsAnalog Info Follows:Region Tone is set for northamericaCurrently processing VoiceMaintenance Mode Set to None (not in mtc mode)Number of signaling protocol errors are 0Impedance is set to 600r OhmAnalog interface A-D gain offset = -3 dBAnalog interface D-A gain offset = -3 dBVoice card specific Info Follows:Signal Type is loopStartRing Frequency is 20 HzHook Status is On HookRing Active Status is inactiveRing Ground Status is inactiveTip Ground Status is activeDigit Duration Timing is set to 100 msInterDigit Duration Timing is set to 100 msRing Cadence are [20 40] * 100 msecInterDigit Pulse Duration Timing is set to 500 msThe following is sample output from the show voice port summary command for all voice ports on a Cisco MC3810 with an analog voice module (AVM):
router# show voice port summaryIN OUT ECHOPORT SIG-TYPE ADMIN OPER IN-STATUS OUT-STATUS CODEC VAD GAIN ATTN CANCEL1/1 fxs-ls up up on-hook idle 729a n 0 0 y1/2 fxs-ls up up on-hook idle 729a n 0 0 y1/3 e&m-wnk up up idle idle 729a n 0 0 y1/4 e&m-wnk up up idle idle 729a n 0 0 y1/5 fxo-ls up up idle on-hook 729a n 0 0 y1/6 fxo-ls up up idle on-hook 729a n 0 0 yexplains the fields in the sample output.
The following is sample output from the Cisco AS5800 for the show voice port command:
5800# show voice port 1/0/0:DISDN 1/0/0:DType of VoicePort is ISDNOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is ""Noise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 16 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sRegion Tone is set for USexplains the fields in the sample output.
Related Commands
show vrm active_calls
To display active-only voice calls either for a specific VFC or all VFCs, use the show vrm active_calls privileged EXEC command.
show vrm active_calls {dial-shelf-slot-number | all}
Syntax Description
dial shelf slot number
Slot number of the dial shelf. Valid number is 0 to 13.
all
Lists all active calls for VFC slots.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use the show vrm active_calls to display active-only voice calls either for a specific VFC or all VFCs. Each active call occupies a block of information describing the call. This information provides basically the same information as the show vrm vdevice command.
Examples
The following is sample output from the show vrm active_calls command specifying dial shelf slot number:
5800# show vrm active_calls 6slot = 6 virtual voice dev (tag) = 61 channel id = 2capabilities list map = 9FFFlast/current codec loaded/used = NoneTDM timeslot = 241Resource (vdev_common) status = 401 means :active otherstot ingress data = 24tot ingress control = 1308tot ingress data drops = 0tot ingress control drops = 0tot egress data = 22051tot egress control = 1304tot egress data drops = 0tot egress control drops = 0slot = 6 virtual voice dev (tag) = 40 channel id = 2capabilities list map = 9FFFlast/current codec loaded/used = NoneTDM timeslot = 157Resource (vdev_common) status = 401 means :active othersexplains the fields in the sample output.
Related Commands
show vrm vdevices
Displays detailed information for a specific DSP or a brief summary display for all VFCs.
show vrm vdevices
To display detailed information for a specific DSP or a brief summary display for all VFCs, use the show vrm vdevices privileged EXEC command.
show vrm vdevices {{vfc-slot-number | voice-device-number} | summary}
Syntax Description
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use the show vrm vdevice to display detailed information for a specific DSP or a brief summary display for all VFCs. The display provides information on the number of channels, channels per DSP, bitmap of DSPMs, version numbers, and so on. This information is useful in monitoring the current state of your VFCs.
The display for a specific DSP provides information on the codec that each channel is using, if active, or last used and if the channel is not currently transmitting cells. It also displays the state of the resource. In most cases, if there is an active call on that channel, the resource should be marked active. If the resource is marked as reset and/or bad, this may be an indication of a response loss for the VFC on a reset request. If this condition persists, you might experience a problem with the communication link between the router shelf and the VFC.
Examples
The following is sample output from the show vrm vdevice command specifying dial shelf slot number and DSP number. In this particular example, the call is active so the statistics displayed are for this active call. If no calls are currently active on the device, the statistics would be for the previous (or last active) call.
5800# show vrm vdevices 6 1slot = 6 virtual voice dev (tag) = 1 channel id = 1capabilities list map = 9FFFlast/current codec loaded/used = NoneTDM timeslot = 0Resource (vdev_common) status = 401 means :active otherstot ingress data = 101tot ingress control = 1194tot ingress data drops = 0tot ingress control drops = 0tot egress data = 39722tot egress control = 1209tot egress data drops = 0tot egress control drops = 0slot = 6 virtual voice dev (tag) = 1 channel id = 2capabilities list map = 9FFFlast/current codec loaded/used = NoneTDM timeslot = 1Resource (vdev_common) status = 401 means :active otherstot ingress data = 21tot ingress control = 1167tot ingress data drops = 0tot ingress control drops = 0tot egress data = 19476tot egress control = 1163tot egress data drops = 0tot egress control drops = 0explains the fields in the sample output.
The following is sample output from the show vrm devices command specifying a summary list. In the Voice Device Mapping area, the C_Ac column indicates number of active calls for a specific DSP. If there are any non zero numbers under the C_Rst and/or C_Bad column, this indicates a reset request was sent but it was lost; this could mean a faulty DSP.
5800# show vrm vdevices summary*****************************************************************summary of voice devices for all voice cards********************************************************************slot = 6 major ver = 0 minor ver = 1 core type used = 2number of modules = 16 number of voice devices (DSPs) = 96chans per vdevice = 2 tot chans = 192 tot active calls = 178module presense bit map = FFFF tdm mode = 1 num_of_tdm_timeslots = 384auto recovery is onnumber of default voice file (core type images) = 2file 0 maj ver = 0 min ver = 0 core_type = 1trough size = 2880 slop value = 0 built-in codec bitmap = 0loadable codec bitmap = 0 fax codec bitmap = 0file 1 maj ver = 3 min ver = 1 core_type = 2trough size = 2880 slop value = 1440 built-in codec bitmap = 40Bloadable codec bitmap = BFC fax codec bitmap = 7E-------------------Voice Device Mapping------------------------Logical Device (Tag) Module# DSP# C_Ac C_Busy C_Rst C_Bad---------------------------------------------------------------1 1 1 2 0 0 02 1 2 2 0 0 03 1 3 2 0 0 04 1 4 2 0 0 05 1 5 2 0 0 06 1 6 2 0 0 0+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++7 2 1 2 0 0 08 2 2 2 0 0 09 2 3 2 0 0 010 2 4 1 0 0 011 2 5 2 0 0 012 2 6 1 0 0 0+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++<information deleted>+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++91 16 1 2 0 0 092 16 2 2 0 0 093 16 3 1 0 0 094 16 4 2 0 0 095 16 5 2 0 0 096 16 6 2 0 0 0+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++Total active call channels = 178Total busied out channels = 0Total channels in reset = 0Total bad channels = 0Note :Channels could be in multiple statesexplains the fields in the sample output.
Related Commands
show vrm active_calls
Displays active-only voice calls either for a specific VFC or all VFCs.
test vrm busyout
To busyout a specific DSP or channels on a specific DSP, use the test vrm busyout privileged EXEC command.
test vrm busyout slot-number {first-dsp-number {last-dsp-number | {channel number}} | all
Syntax Description
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use the test vrm busyout command to busy out either one specific DSP or a range of DSPs on a specific VFC. In addition, you can use this comand to busyout a particular channel on a specified DSP or range of DSPs. To restore the activity of the busied-out DSP(s), use the test vrm unbusyout command.
Examples
The following example busies out all of the DSPs and associated channels for the VFC located in slot 4:
router# test vrm busyout 4 allThe following example busied out all of the channels from DSP1 to DSP3 for the VFC located in slot 4:
router# test vrm busyout 4 1 3The following example busies out only channel 2 of DSP1 for the VFC located in slot 4:
router# test vrm busyout 4 1 channel 2Related Commands
test vrm unbusyout
Restores activity to a busied-out DSP or busied-out channels on a DSP.
test vrm reset
To reset a particular DSP, use the test vrm reset privileged EXEC command.
test vrm reset {slot-number dsp-number}
Syntax Description
slot-number
Number identifing the slot where the VFC is installed.
dsp-number
Number identifying the DSP to be reset.
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use the test vrm reset command to send a hard reset command to an identified DSP. When this command is used, any active calls on all channels associated with this DSP are dropped. Under most circumstances, you will never need to use this command.
Examples
The following example resets DSP 4 on the VFC installed in slot 2:
router# test vrm reset 4 2Resetting voice device may termiate active calls [confirm}Reset command sent to voice card 4 for voice device 2.test vrm unbusyout
To restore activity to a busied-out DSP or busied-out channels on a DSP, use the test vrm unbusyout privileged EXEC command.
test vrm unbusyout slot-number {first-dsp-number {last-dsp-number | {channel number}} | all
Syntax Description
Defaults
No default behavior or values.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use the test vrm unbusyout command to restore either one specific DSP or a range of DSPs on a specific VFC. In addition, you can use this comand to restore a particular channel on a specified DSP or range of DSPs. To busy out a DSP (or range of DSPs) or to busy out a particular channel, use the test vrm busyout command.
Examples
The following example restores the activity of all of the DSPs and associated channels for the VFC located in slot 4:
router# test vrm unbusyout 4 allThe following example restores the activity of all the channels on the DSP from DSP1 to DSP3 for the VFC located in slot 4:
router# test vrm unbusyout 4 1 3The following example restores the activity of only channel 2 of DSP1 for the VFC located in slot 4:
router# test vrm unbusyout 4 1 channel 2Related Commands
voice-port
To enter the voice-port configuration mode, use the voice-port global configuration command.
Cisco 2600/3600 Series Router
voice-port slot-number/subunit-number/port
Cisco MC3810
voice-port [slot/port] [summary]
Cisco AS5300 Access Router
voice-port controller number:D
Cisco AS5800 Universal Access Router
voice-port {shelf/slot/port:D} | {shelf/slot/parent:port:D}
Syntax Description
For the Cisco 2600/3600 series:
For the Cisco MC3810:
For the Cisco AS5300 Access Server:
controller number
Specifies the T1 or E1 controller.
:D
Indicates the D channel associated with ISDN PRI.
For the Cisco AS5800 Universal Access Server:
Defaults
No default behavior or values.
Command Modes
Global configuration
Command History
Usage Guidelines
Use the voice-port global configuration command to switch to the voice-port configuration mode from the global configuration mode. Use the exit command to exit the voice-port configuration mode and return to the global configuration mode.
Examples
The following example accesses the voice-port configuration mode for port 0, located on subunit 0 on a voice interface card installed in slot 1 for the Cisco 3600 series:
configure terminalvoice-port 1/0/0The following example accesses the voice-port configuration mode for digital voice port 24 on a Cisco MC3810 with a DVM installed:
configure terminalvoice-port 1/24The following example accesses the voice-port configuration mode for the Cisco AS5300:
configure terminalvoice-port 1:DThe following example accesses the voice-port configuration mode for the Cisco AS5800 (T1 card):
configure terminalvoice-port 1/0/0:DThe following example accesses the voice-port configuration mode for the Cisco AS5800 (T3 card):
configure terminalvoice-port 1/0/0:1:DRelated Commands
dial-peer voice
Enters dial-peer configuration mode and specifies a tag number for a dial peer.
Debug Commands
This section documents new or modified debug commands. All other commands used with this feature are documented in one of the follwing Cisco IOS documentation:
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New Debug Commands
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debug vrm control
To display all control messages sent to and received from the DSP, use the debug vrm control privileged EXEC command. To stop displaying DSP-specific control messages, use the no form of this command.
[no] debug vrm control
Syntax Description
There are no arguments or keywords used in this command.
Defaults
No default behavior or values.
Command History
Examples
The following example displays DSP-specific control messages going to the VRM:
*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size C*Nov 22 19:17:49.351: content : 0 0 0 1 0 8 0 1 0 4B 0 0 0 0 0*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size 14*Nov 22 19:17:49.351: content : 0 0 0 1 0 10 0 1 0 4A 0 1 0 0 0*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size 1C*Nov 22 19:17:49.351: content : 0 0 0 1 0 18 0 1 0 5C 0 2 0 2 0*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size 16*Nov 22 19:17:49.351: content : 0 0 0 1 0 12 0 1 0 4C 0 3 0 1 0*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size E*Nov 22 19:17:49.351: content : 0 0 0 1 0 A 0 1 0 42 0 4 0 0 0*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size 10*Nov 22 19:17:49.351: content : 0 0 0 1 0 C 0 1 0 5B 0 5 0 0 0*Nov 22 19:17:49.351: SEND CONTROL slot 4 tag 1 size E*Nov 22 19:17:49.351: content : 0 0 0 1 0 A 0 1 0 4E 0 6 FF DA 0*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size C*Nov 22 19:17:51.995: content : 0 0 0 1 0 8 0 1 0 44 0 7 FF DA 0*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size C*Nov 22 19:17:51.995: content : 0 0 0 1 0 8 0 1 0 47 0 8 FF DA 0*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size C*Nov 22 19:17:51.995: content : 0 0 0 1 0 8 0 1 0 44 0 9 FF DA 0*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size 1C*Nov 22 19:17:51.995: content : 0 0 0 1 0 18 0 1 0 5C 0 A 0 2 0*Nov 22 19:17:51.995: SEND CONTROL slot 4 tag 1 size 1C*Nov 22 19:17:51.995: content : 0 0 0 1 0 18 0 1 0 49 0 B 0 1 0*Nov 22 19:17:54.815: SEND CONTROL slot 4 tag 1 size E*Nov 22 19:17:54.815: content : 0 0 0 1 0 A 0 1 0 53 0 1 0 0 0*Nov 22 19:17:54.815: SEND CONTROL slot 4 tag 1 size E*Nov 22 19:17:54.815: content : 0 0 0 1 0 A 0 1 0 54 0 1 0 0 0*Nov 22 19:17:54.815: SEND CONTROL slot 4 tag 1 size E*Nov 22 19:17:54.815: content : 0 0 0 1 0 A 0 1 0 57 0 1 0 0 0*Nov 22 19:17:54.827: nip_voice_service_cb : Msg from DS slot 4 cmd = 196.*Nov 22 19:17:54.827: RECEIVED CONTROL slot 4 tag 1 size 1C*Nov 22 19:17:54.827: content : 0 0 0 1 0 18 0 1 0 C4 0 1 8F EA 9B*Nov 22 19:17:54.827: DSP msg 196 received*Nov 22 19:17:54.827: nip_voice_service_cb : Msg from DS slot 4 cmd = 197.*Nov 22 19:17:54.827: RECEIVED CONTROL slot 4 tag 1 size 24*Nov 22 19:17:54.827: content : 0 0 0 1 0 20 0 1 0 C5 0 1 0 0 0*Nov 22 19:17:54.827: DSP msg 197 received*Nov 22 19:17:54.827: nip_voice_service_cb : Msg from DS slot 4 cmd = 200.*Nov 22 19:17:54.827: RECEIVED CONTROL slot 4 tag 1 size 34*Nov 22 19:17:54.827: content : 0 0 0 1 0 30 0 1 0 C8 0 1 0 0 0*Nov 22 19:17:54.827: DSP msg 200 received*Nov 22 19:17:58.539: SEND CONTROL slot 4 tag 1 size E*Nov 22 19:17:58.539: content : 0 0 0 1 0 A 0 1 0 53 0 1 0 0 0*Nov 22 19:17:58.539: SEND CONTROL slot 4 tag 1 size E*Nov 22 19:17:58.539: content : 0 0 0 1 0 A 0 1 0 54 0 1 0 0 0*Nov 22 19:17:58.539: SEND CONTROL slot 4 tag 1 size E*Nov 22 19:17:58.539: content : 0 0 0 1 0 A 0 1 0 57 0 1 0 0 0*Nov 22 19:17:58.551: nip_voice_service_cb : Msg from DS slot 4 cmd = 196.*Nov 22 19:17:58.555: RECEIVED CONTROL slot 4 tag 1 size 1C*Nov 22 19:17:58.555: content : 0 0 0 1 0 18 0 1 0 C4 0 1 8F EA 9B*Nov 22 19:17:58.555: DSP msg 196 received*Nov 22 19:17:58.555: nip_voice_service_cb : Msg from DS slot 4 cmd = 197.*Nov 22 19:17:58.555: RECEIVED CONTROL slot 4 tag 1 size 24*Nov 22 19:17:58.555: content : 0 0 0 1 0 20 0 1 0 C5 0 1 0 0 0*Nov 22 19:17:58.555: DSP msg 197 received*Nov 22 19:17:58.555: nip_voice_service_cb : Msg from DS slot 4 cmd = 200.*Nov 22 19:17:58.555: RECEIVED CONTROL slot 4 tag 1 size 34*Nov 22 19:17:58.555: content : 0 0 0 1 0 30 0 1 0 C8 0 1 0 0 0*Nov 22 19:17:58.555: DSP msg 200 received*Nov 22 19:18:02.127: SEND CONTROL slot 4 tag 1 size C*Nov 22 19:18:02.127: content : 0 0 0 1 0 8 0 1 0 47 0 C 0 0 0Format of the Send messages is as follows:
SEND CONTROL slot <slot#> tag <tag#> size <size>content : <x x x x> <x x> <x x> <x x> <x x> <x x x>tag# len chan msg proc rtp_headerFormat for the Receive messages is as follows:
nip_voice_service_cb : Msg from DS slot <slot#> cmd = <msg>.RECEIVED CONTROL slot <slot#> tag <tag#> size <size>content : 0 0 0 1 0 18 0 1 0 C4 0 1 8F EA 9Bcontent : <x x x x> <x x> <x x> <x x> <x x> <x x x>tag# len chan msg proc rtp_headerDSP msg <msg> receiveddescribes the fields in previous example.
Related Commands
debug vrm error
To display all DSP-specific error messages going to the voice resource manager (VRM), use the debug vrm error privileged EXEC command. To stop displaying DSP-specific error messages, use the no form of this command.
[no] debug vrm error
Syntax Description
There are no arguments or keywords used in this command.
Defaults
No default behavior or values.
Command History
Examples
The following examples show some possible outputs from the debug vrm error command, displaying DS_specific error messages.
This example shows that an error occurred when sending data from the DSP to IP network (ingress direction):
- vrm_vtsp_send_ingress_data : fs_input failedThis error message shows that an error occurred when sending control message from the DSP to VTSP:
- vrm_vtsp_send_ingress_control : failedThis error message shows that there is no voice card present and a voice call is attempted:
- vrm_vtsp: No Voice Card ready yet.This error message shows that no free resource is available, and a voice call is attempted:
- vrm_vtsp_open : vdev_common not availableThis error message shows that there is already an active call on this channel, so abort:
- vrm_vtsp_open : vchan_instance already in use ABORT OPENThe following messages show that the VTSP did a "dirty close" on a particular channel. "Dirty close" means that the DSP did not respond to the VTSP's request for the final statistics of the call.
- vrm_vtsp_open : cdb->dsp_info not NULL Abort OPEN- vrm_vtsp_close failure no vtsp_cdb_ptr- vrm_vtsp_close: without a dsp_info!- vrm_vtsp_close : dirty close on tag <tag#> channel <chan#>The following error mesaage describes the status of the DSP (virtual device):
- vrm_vtsp_close : vdev freed not locked. Status <value>Possibe status values are as follows:
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The following error message shows that a "set_codec" command was issued, but the codec was not supported by the DSP:
- VTSP_FAIL: codec <value> not supportedPossible codec values are as follows:
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This error message shows that there is no buffer left in the pool for the VTSP to send a message to the DSP. <Number> int his output referst o the number of times the VRM ran out of buffer space.
- vrm_vtsp_get_packet: no buffers <number>This error message notifies the VRM of a DSP alarm:
- vrm_vtsp_indicate_alarm : alarm_type <value> slot <slot#> tag <tag#> chan <chan#>Possible values for the alarm are as follows:
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This eror message shows that the DSP sent a defective message:
- vrm msg offset too big tag <tag#> vchan <chan#>expains the field contained in the previous example.
This error message indicates that an alarm message was received from the VFC/DSP and was successfully sent to the VTSP:
- vrm_msg_process_alarm_msg for <slot#>.<tag#>.<chan#> , state=<value>Possible state values are as follows:
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Related Commands
debug vrm inout
To display debug messages for all DSP-specific messages going to and coming from the voice resource manager (VRM), use the debug vrm inout privileged EXEC command. To stop displaying DSP-specific messages, use the no form of this command.
[no] debug vrm inout
Syntax Description
There are no arguments or keywords used in this command.
Defaults
No default behavior or values.
Command History
Examples
The following example displays DSP-specific messages going to the VRM when a call is made:
*Jun 17 13:02:41.495:vrm_vtsp_open :vtsp_cdb_ptr 623D2170*Jun 17 13:02:41.495:vrm_vtsp_open :VTSP_SUCCESS*Jun 17 13:02:41.535:vrm_vtsp_get_capabilities :vtsp_cdb_ptr 623D2170*Jun 17 13:02:41.535:vrm_vtsp_get_capabilities :vtsp_cdb_ptr 623D2170*Jun 17 13:02:41.535:vrm_vtsp_set_codec :vtsp_cdb_ptr 623D2170 new_codec 5*Jun 17 13:02:41.535:VTSP_SUCCESS:Codec 5 was loaded already.*Jun 17 13:02:41.767:vrm_vtsp_set_codec :vtsp_cdb_ptr 623D2170 new_codec 5*Jun 17 13:02:41.767:VTSP_SUCCESS:Codec 5 was loaded already.The following example displays DSP-specific messages going to the VRM when a call is complete:
*Jun 17 13:02:49.119:vrm_vtsp_close :vtsp_cdb_ptr 623D2170*Jun 17 13:02:49.119:vrm_vtsp_close :0x2 close OKRelated Commands
Glossary
AAA—Authentication, Authorization, and Accounting. AAA is a suite of network security services that provide the primary framework through which access control can be set up on your Cisco router or access server.
ACOM—Term used in G.165, "General Characteristics of International Telephone Connections and International Telephone Circuits: Echo Cancellers." ACOM is the combined loss achieved by the echo canceller, which is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call.
a-law—A voice compression technique commonly used in Europe.
ANI—Answer Number Indication. The calling number (number of calling party).
ARQ—Admission request.
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. In E1 applications, timeslot 16 is used to transmit CAS information. Each frame's timeslot 16 carries signaling information (ABCD bits) for two of the B channel timeslots.
CIR—Committed Information Rate. The average rate of information transfer a subscriber (for example, the network administrator) has stipulated for a Frame Relay PVC.
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 IP, it specifies the voice coder rate of speech for a dial peer.
Data Link Connection Identifier (DLCI)—Frame Relay virtual circuit number corresponding to a particular destination. The DLCI is part of the Frame Relay header and is usually 10 bits long.
Dial peer—An addressable call endpoint. In Voice over IP, there are two kinds of dial peers: POTS and VoIP.
DNS—Domain Name System used to address translation to convert H.323 IDs, URLs, or e-mail IDs to IP addresses. DNS is also used to assist in the location of remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains.
DNIS—Dialed number identification service. The destination number.
DS0—A 64-Kbps channel on an E1 or T1 WAN interface.
DSP—Digital Signal Processor.
DTMF—Dual tone multifrequency. Use of two simultaneous voice-band tones for dial (such as touch tone).
E.164—The international public telecommunications numbering plan. A standard set by ITU-T which addresses telephone numbers.
E1—Wide-area digital transmission scheme. E1 is the European equivalent of a T1 line. The E1's higher clock rate (2.048 MHz) allows for 32 64-Kbps channels, which include one channel for framing and one channel for D-channel information.
E&M—Ear and mouth RBS signaling.
Endpoint—An H.323 terminal or gateway. An endpoint can call and be called. It generates and/or terminates the information stream.
FIFO—First-in, first-out. In data communication, FIFO refers to a buffering scheme where the first byte of data entering the buffer is the first byte retrieved by the CPU. In telephony, FIFO refers to a queuing scheme where the first calls received are the first calls processed.
Gatekeeper—A gatekeeper maintains a registry of devices in the multimedia network. The devices register with the gatekeeper at startup, and request admission to a call from the gatekeeper.
The gatekeeper is an H.323 entity on the LAN that provides address translation and control access to the LAN for H.323 terminals and gateways. The gatekeeper may provide other services to the H.323 terminals and gateways, such as bandwidth management and locating gateways.
Gateway—A gateway allows H.323 terminals to communicate with non-H.323 terminals by converting protocols. A gateway is the point at which a circuit-switched call is encoded and repackaged into IP packets.
An H.323 gateway is an endpoint on the LAN that provides real-time two-way communications between H.323 terminals on the LAN and other ITU-T terminals in the WAN, or to another H.323 gateway.
H.323—An International Telecommunication Union (ITU-T) standard that describes packet-based video, audio, and data conferencing. H.323 is an umbrella standard that describes the architecture of the conferencing system, and refers to a set of other standards (H.245, H.225.0, and Q.931) to describe its actual protocol.
H.323 RAS—Registration, admission, and status. The RAS signaling function performs registration, admissions, bandwidth changes, status and disengage procedures between the VoIP gateway and the gatekeeper.
HSRP—Hot Standby Routing Protocol. HSRP is a Cisco proprietary protocol which provides a redundancy mechanism when more than one router is connected to the same segment/subnet of an Ethernet/FDDI/Token Ring network.
ISDN—Integrated Services Digital Network. ISDN is a communications protocol, offered by telephone companies, that permits telephone networks to carry data, voice, and other traffic.
ITU-T—Telecommunication standardization sector of ITU.
IVR—Integrated voice response. A software feature that allows the use of one of several interactive voice response scripts during the call processing functionality.
LEC—Local exchange carrier.
LRQ—Location request.
MCU—Multipoint control unit
mu-law—a-law—A voice compression technique commonly used in North America.
Multicast—A process of transmitting PDUs from one source to many destinations. The actual mechanism (that is, IP multicast, multi-unicast, etc.) for this process may be different for LAN technologies.
Multilink PPP—Multilink Point-to-Point Protocol. This protocol is a method of splitting, recombining, and sequencing datagrams across multiple logical data links.
Multipoint-unicast—A process of transferring Protocol Data Units (PDUs) where an endpoint sends more than one copy of a media stream to different endpoints. This may be necessary in networks which do not support multicast.
node—An H.323 entity that uses RAS to communicate with the gatekeeper. For example, an endpoint such as a terminal, proxy, or gateway.
PDU—Protocol Data Units. Used by bridges to transfer connectivity information.
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.
PRI—Primary Rate Interface. PRI is an ISDN interface to primary rate access. Primary rate access consists of a single 64 Kbps D channel plus 23 T1 or 30 E1 B channels for voice or data.
PSTN—Public Switched Telephone Network. PSTN refers to the local telephone company.
PVC—Permanent Virtual Circuit.
QoS—Quality of Service, which refers to the measure of service quality provided to the user.
RAS—Registration, Admission, and Status Protocol. This is the protocol that is used between endpoints and the gatekeeper to perform management functions.
RBS—Robbed Bit Signaling
RRQ—Registration request.
RSVP—Resource Reservation Protocol. This protocol supports the reservation of resources across an IP network.
T1—Digital WAN carrier facility. T1 transmits DS-1 formatted data at 1.544 Mbps through the telephone-switching network, using AMI or B8ZS coding. T1 is the North American equivalent of an E1 line.
TCL—Tool Command Language. An interpreted script language developed by Dr. John Ousterhout of the University of California, Berkeley, and now developed and maintained by Sun Microsystems Laboratories.
U-law—A companding technique commonly used in North America. U-law is standardized as a 64-Kbps codec in ITU-T G.711.
SPI—Service provider interface.
TDM—Time division multiplexing. Technique in which information from multiple channels can be allocated bandwidth on a single wire based on preassigned time slots. Bandwidth is allocated to each channel regardless of whether the station has data to transmit.
VoIP—Voice over IP. The ability to carry normal telephone-style voice over an IP-based internet with POTS-like functionality, reliability, and voice quality. VoIP is a blanket term which generally refers to Cisco's standards based (H.323, etc.) approach to IP voice traffic.
VoIP dial peer—Dial peer connected via a packet network; in the case of Voice over IP, this is an IP network. VoIP peers point to specific VoIP devices.
VTSP—Voice telephony service provider.
Zone—A collection of all terminals (tx), gateways (GW), and Multipoint Control Units (MCU) managed by a single gatekeeper (GK). A Zone includes at least one terminal, and may or may not include gateways or MCUs. A Zone has only one gatekeeper. A Zone may be independent of LAN topology and may be comprised of multiple LAN segments which are connected using routes or other devices.
