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Table Of Contents
Integrated Data, Voice, and Video Services for ISDN Interfaces
Prerequisites for Configuring Integrated Data, Voice, and Video Services for ISDN Interfaces
Restrictions for Configuring Integrated Data, Voice, and Video Services for ISDN Interfaces
Information About Integrated Data, Voice, and Video Services for ISDN Interfaces
Primary and Secondary Incoming H.320 Calls
Dynamic and Static H.320 Secondary Called Numbers
How to Configure Integrated Data, Voice, and Video Services for ISDN Interfaces
Enabling Integrated Services on the Interface
Configuring ISDN Inbound POTS Dial Peers
Configuring the Voice Class Codec
Configuring the VoIP Dial Peer
How to Configure Static and Dynamic H.320 Secondary Call Dial Plans
Configuring Dynamic H.320 Secondary Call Dial Plans
Defining Voice Class Called Number Pool for Dynamic Dial Plan
Configuring Dynamic Dial Plan Inbound POTS Dial Peer for Terminating Gateway
Configuring Called Number Pool on Voice Port
Configuring Dynamic Dial Plan Outbound POTS Dial Peer for Originating Gateway
Configuring Static H.320 Secondary Call Dial Plans
Defining Inbound Voice Class Called Numbers for Static Dial Plan
Defining Outbound Voice Class Called Numbers for Static Dial Plan
Configuring Static Dial Plan Outbound POTS Dial Peer for Originating Gateway
Configuring Static Dial Plan Inbound POTS Dial Peer for Terminating Gateway
Configuring a Combined Static and Dynamic H.320 Secondary Call Dial Plan
Configuring the Outbound Static Called Numbers for Combined Static and Dynamic Dial Plan
Configuring Combined Static and Dynamic Dial Plan Inbound POTS Dial Peer for Originating Gateway
Configuring Dynamic Outbound POTS Dial Peers for Terminating Gateway
Configuring Static Outbound POTS Dial Peers for Terminating Gateway
Configuration Examples for Integrated Data, Voice, and Video Services for ISDN Interfaces
Integrated Services with Combined Static and Dynamic H.320 Secondary Call Dial Plan: Example
Integrated Services with Static H.320 Secondary Call Dial Plan: Example
show voice class called-number
show voice class called-number-pool
voice-class called-number (dial peer)
voice-class called-number-pool
Feature Information for Integrating Data, Voice, and Video for ISDN Interfaces
Integrated Data, Voice, and Video Services for ISDN Interfaces
First Published: November 17, 2006, OL-10383-01Last Revised: November 4, 2009The Integrated Data, Voice, and Video Services for ISDN Interfaces feature allows multimedia communications between H.320 endpoints and H.323 or Skinny Client Control Protocol (SCCP) endpoints.
Your software release may not support all the features documented in this module. For the latest feature information and caveats, see the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the "Feature Information for Integrating Data, Voice, and Video for ISDN Interfaces" section.
Use Cisco Feature Navigator to find information about platform support and Cisco IOS and Catalyst OS software image support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
Contents
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Prerequisites for Configuring Integrated Data, Voice, and Video Services for ISDN Interfaces
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Prerequisites for Configuring Integrated Data, Voice, and Video Services for ISDN Interfaces
Before you configure integrated services using H.320 protocol, you must do the following:
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Supported Routers, Hardware Modules, Codecs, Endpoints, and Topologies
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Supported Topologies
Integrated services for ISDN BRI and PRI interfaces allows multimedia communications between H.320 endpoints and ITU-T Recommendation H.323 or SCCP endpoints, including the following topologies:
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H.323 endpoint > H.320 gateway > BRI or PRI interface > H.320 endpoint
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SCCP endpoint > H.320 gateway > BRI or PRI interface > H.320 endpoint
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SCCP endpoint > H.320 gateway > BRI or PRI interface > H.320 gateway > SCCP endpoint
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H.323 endpoint > H.320 gateway > BRI or PRI interface > H.320gateway > H.323 endpoint
Restrictions for Configuring Integrated Data, Voice, and Video Services for ISDN Interfaces
Restrictions for configuring integrated services for ISDN interfaces are as follows:
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Information About Integrated Data, Voice, and Video Services for ISDN Interfaces
Integrated data, voice, and video services through a single ISDN interface allows multimedia communications between H.320 endpoints and H.323 or SCCP endpoints. Before you configure integrated services for ISDN interfaces, you should be familiar with the following concepts:
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Integrated Services Mode
An ISDN interface must be configured for integrated services mode to enable H.320 primary and secondary call type checking. Enabling integrated services allows data, voice, and video call traffic to occur from a single ISDN BRI or PRI interface. When an interface is in integrated service mode:
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By default, the integrated services option is disabled from the supported interfaces.
Primary and Secondary Incoming H.320 Calls
An H.320 call consists of 1 to 16 ISDN B-channels. The first B-channel in an H.320 session is the primary B-channel and all additional B-channels are handled as secondary B-channels. Secondary B-channels are distinguished from primary B-channels by the call number received in the Q.931 ISDN setup message. The secondary called numbers for H.320 B-channels can be exchanged between the terminals using H.242 format (dynamic method), or can be configured statically (static method).
An H.320 primary B-channel is different from the secondary B-channels in the following ways:
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During inbound dial peer matching, the list of H.320 sessions is searched before the incoming voice dial peer lookup. If the new called number matches a called number associated with an existing H.320 call session (dynamic or static), the leg is added to the existing H.320 call session as a secondary B-channel.
The B-channel aggregator is responsible for handling call setup of additional B-channels for H.320 calls. It also allocates dynamic called numbers from the voice class called number pool to the gateway and frees them back up again.
The B-channel aggregator creates a video conferencing session for individual incoming H.320 primary calls. The setup and teardown of each B-channel is handled as one independent call on the ISDN side, which means that each H.320 call can have multiple B-channels. On the H.323 side, only one call is presented to the endpoint. For this reason, multiple ISDN calls are grouped together to form one logical H.320 to H.323 call. The H.320 B-channel aggregator provides this function.
Dynamic and Static H.320 Secondary Called Numbers
A called number is a digit string that can be matched by an incoming or outgoing call to associate the call with a dial peer. From the originating gateway, a set of unique incoming called numbers can be allocated for an incoming H.320 primary call to the originating H.320 terminal. The allocated incoming called numbers are associated with one active H.320 session and used by the originating H.320 terminal as dialing numbers to initiate the H.320 secondary calls.
To connect secondary B-channels into an H.320 call, additional called numbers might be needed if each leg has a called number different from the primary. This is accomplished using either dynamic or static secondary dial plans.
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Dynamic Called Numbers
A called number pool is a group of dynamic called numbers to be referenced by the gateway for handling primary and secondary calls. If the originating H.320 terminal supports receiving dynamic secondary called numbers (H.242), the H.320 leg aggregator module allocates the idle called numbers from a pool referenced by the voice interface on the originating gateway for the H.320 primary call. The number of dynamic called numbers to be allocated is based on the bandwidth requirement of the incoming H.320 session.
Static Called Numbers
Static called numbers are configured for H.320 endpoints that are not capable of receiving dynamic secondary calling numbers (non-H.242). The static called numbers are referenced by the incoming and outgoing POTS dial peers. Up to 15 called numbers (in E.164 format) can be configured as static called numbers to match the incoming H.320 secondary calls.
Video Information Type
When a dial peer is created, the default information type is voice. To enable H.320 call support, you must configure a video information type on the POTS dial peer for inbound dial peer matching.
A POTS dial peer configured with a video information-type is marked as a specific type of voice dial peer. During the ISDN call type checking for an incoming H.320 call, the matching of voice dial peers with video information-type takes precedence over the matching of voice dial peers with other information type settings. Outgoing H.320 primary calls are initiated by the default application by matching an outbound POTS dial peer with a video information type.
An incoming POTS dial peer with a video information type provisions for incoming H.320 primary calls using the incoming called-number.
Bandwidth for H.320 Calls
Each c5510 digital signal processor (DSP) channel supports 64 kilobits of bandwidth. Each c5510 DSP has 16 channels available. One of those channels can support a bandwidth of 1024 kbps, allowing the DSP to support one H.320 call with a maximum of 16 B-channels. For each dial peer configured for information-type video, an optional bandwidth command can be added that specifies the minimum acceptable and maximum allowed bandwidth for the H.320 call, in 64-kbit increments. If the number of call legs connected falls between the minimum and maximum configured, then video is allowed. If the minimum bandwidth cannot be met for the call, the call drops back to an audio-only H.320 call.
How to Configure Integrated Data, Voice, and Video Services for ISDN Interfaces
This section describes how to configure integrated data, voice, and video services for ISDN BRI or PRI interfaces, and includes the following tasks:
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Enabling Integrated Services on the Interface
Enabling integrated services allows video and voice call traffic to occur from ISDN BRI or PRI interfaces simultaneously.
When an interface is in integrated service mode:
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By default, the integrated service option is disabled from the supported interfaces. Use the following procedure to enable integrated mode on a serial interface.
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Examples
In the following example, the interface is shut down.
Router(config)# interface Serial4/1:15Router(config-if)# shutdownThis example shows that integrated mode is enabled.
Router(config)# interface Serial4/1:15Router(config-if)# isdn integrate calltype all% This command line will enable the Serial Interface to "integrated service" mode.% The "isdn incoming-voice voice" setting will be removed from the interface.% Continue? [confirm]When you confirm, the default incoming-voice configuration is removed from the interface, and the interface is now in integrated service mode. The interface does not reset back to voice mode if an incoming call is originated from the interface.
This example show the interface being set to active again.
Router(config)# interface Serial4/1:15Router(config-if)# no shutdownConfiguring ISDN Inbound POTS Dial Peers
Use the following procedure to configure the inbound POTS dial peer for an ISDN interface.
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Examples
dial-peer voice 12 potsinformation-type videoincoming called-number 408direct-inward-dialTroubleshooting Tips
Use the show dial-peer voice command to verify the dial peer configuration.
What to Do Next
To configure a voice class codec, continue with the "Configuring the Voice Class Codec" section. If a voice class codec is already configured, or if you plan reference a video codec on the dial peer, proceed to the "Configuring the VoIP Dial Peer" section.
Configuring the Voice Class Codec
Use this procedure to configure a voice class codec, to be referenced by the VoIP dial peer.
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Multiple video codecs can be defined to a voice class codec, as shown in the following example.
voice class codec 10codec preference 1 g722codec preference 2 g711alawvideo codec h261video codec h263video codec h264Configuring the VoIP Dial Peer
Use the following procedure to configure the inbound or outbound VoIP dial peer.
Restrictions
Restrictions for configuring the VoIP dial peer are as follows:
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destination pattern [+] string [T] (outgoing dial peer)
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video codec [h261 | h263 | h263+ | h264]
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Examples
dial-peer voice 12 voipdestination-pattern 4085550100video codec h263+rtp payload-type 118dial-peer voice 12 voipshutdownincoming called-number 408voice-class codec 10Troubleshooting Tips
Use the show dial-peer voice command to verify the dial peer configuration.
What to Do Next
Configure a secondary call dial plan, for both H.242 (dynamic) and nonH.242 endpoints (static) using one or more of the following sections.
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How to Configure Static and Dynamic H.320 Secondary Call Dial Plans
If your endpoint is capable of dynamic receipt of secondary calling numbers (using H.242), configure a dynamic H.320 secondary call dial plan. To configure the secondary call number statically (nonH.242 endpoints), configure a static H.320 secondary call dial plan.
This section describes how to configure static and dynamic H.320 secondary call dial plans and includes the following tasks:
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Configuring Dynamic H.320 Secondary Call Dial Plans
Use a dynamic secondary call dial plan when a gateway is connected to a H.320 endpoint that supports dynamic allocation of secondary call numbers (using H.242).
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Use the following tasks to configure a dynamic H.320 secondary call dial plan.
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Defining Voice Class Called Number Pool for Dynamic Dial Plan
In a dynamic dial plan, you define a pool of dynamic called numbers to be referenced by the gateway for handling primary and secondary calls. Use the following procedure to configure a voice class called number pool for the dynamic H.320 secondary call dial plan.
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voice class called number pool 100index 1 6505550100 - 6505550111
voice class called number pool 200index 1 6505550100 - 6505550111 (Range of called numbers are 6505550100 up to 6505550111)index 2 6505550112 - 6505550121 (Range of called numbers are 6505550112 up to 6505550121)Configuring Dynamic Dial Plan Inbound POTS Dial Peer for Terminating Gateway
The dynamic inbound POTS dial peer on the terminating gateway handles outgoing H.320 primary and secondary calls. Define the POTS dial peer with ISDN trunk group as the routing interface. The called number for the outgoing H.320 secondary calls are retrieved from the remote H.320 endpoint.
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Use the following steps to configure an inbound POTS dial peer for a terminating gateway.
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Examples
dial-peer voice 12 potsinformation-type videodestination-pattern 4085550100bandwidth maximum 256 minimum 64no digit-striptrunkgroup isdntgTroubleshooting Tips
Use the show dial-peer voice command to verify the dial peer configuration.
Configuring Called Number Pool on Voice Port
Dynamic called number support for ISDN calls occurs at the voice port level. Multiple ISDN interfaces can reference the same called number pool if the range of dynamic called numbers are valid routing dialed numbers from the H.320 endpoint to the originating gateway. Use the following steps to assign the voice class called number pool to the ISDN voice port.
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voice class called number pool 100index 1050 - 1075dial-peer voice 1000 potsdestination-pattern 1000information-type videobandwidth maximum 1024voice-port 1/0:23voice-class called-number-pool 100Troubleshooting Tips
Use the show voice port command to verify voice port configuration.
Configuring Dynamic Dial Plan Outbound POTS Dial Peer for Originating Gateway
Use the following steps to configure an outbound POTS dial peer on the originating gateway, including the settings for maximum bandwidth and a video information-type.
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dial-peer voice 1000 potsdestination-pattern 1000information-type videobandwidth maximum 1024 minimum 64port 1/0:23Troubleshooting Tips
Use the show dial-peer voice command to verify the dial peer configuration.
What to Do Next
To configure a static H.320 secondary dial plan, proceed to the "Configuring Static H.320 Secondary Call Dial Plans" section. To configure a combined static and dynamic H.320 secondary dial plan, proceed to the "Configuring a Combined Static and Dynamic H.320 Secondary Call Dial Plan" section.
Configuring Static H.320 Secondary Call Dial Plans
Use a static secondary call dial plan when a gateway is connected to a H.320 endpoint that does not support H.242. A static secondary call dial plan uses called number tables in E.164 format to use as called numbers for incoming and outgoing calls to H.320 endpoints.
Use the following tasks to configure a static H.320 secondary call dial plan:
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Defining Inbound Voice Class Called Numbers for Static Dial Plan
Define a the inbound called number table to associate incoming H.320 secondary calls with H.320 primary calls. Use this procedure to define one or more voice class called numbers for the inbound POTS dial peers.
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voice class called number inbound 200index 1 5550100
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index 4 5550103voice class called number inbound 9001index 1 9001!voice class called number inbound 9999index 1 9997index 2 9998index 3 9999!Defining Outbound Voice Class Called Numbers for Static Dial Plan
Define an outbound called number table to associate outgoing H.320 secondary calls with H.320 primary calls. Use these steps to define one or more voice class called number for the outbound POTS dial peers.
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voice class called number outbound 50index 1 100A11 index 2 +7878*55voice class called number outbound 1index 1 6001!voice class called number outbound 7101index 1 7101!voice class called number outbound 1111index 1 1111index 2 1112index 3 1113index 4 1114!Configuring Static Dial Plan Outbound POTS Dial Peer for Originating Gateway
The originating gateway handles outgoing H.320 primary and secondary calls. Use this procedure to configure the outbound POTS dial peer for the originating gateway for a static dial plan, including the outbound called number table.
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dial-peer voice 7001 potsinformation-type videovoice-class called-number inbound 1incoming called-number 408bandwidth maximum 256 minimum 64direct-inward-dialTroubleshooting Tips
Use the show dial-peer voice command to verify the dial peer configuration.
Configuring Static Dial Plan Inbound POTS Dial Peer for Terminating Gateway
The terminating gateway handles incoming H.320 primary and secondary calls. Use this procedure to configure the inbound POTS dial peer for the terminating gateway for a static dial plan, including the inbound called number table.
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dial-peer voice 12 potsinformation-type videovoice-class called-number inbound 50destination-pattern 4085550100bandwidth maximum 192no digit-striptrunkgroup isdntgTroubleshooting Tips
Use the show dial-peer voice command to verify the dial peer configuration.
What to Do Next
To configure a combined static and dynamic H.320 secondary dial plan, proceed to the "Configuring a Combined Static and Dynamic H.320 Secondary Call Dial Plan" section. To configure a dynamic dial plan, proceed to the "Configuring Dynamic H.320 Secondary Call Dial Plans" section.
Configuring a Combined Static and Dynamic H.320 Secondary Call Dial Plan
Determining whether to use static or dynamic H.320 secondary dial plan depends on the capability of the remote H.320 endpoints. In some networks, the ISDN interface between an originating and terminating gateway might need to support both static and dynamic dial plans.
Use the following tasks to configure a combined static and dynamic H.320 secondary call dial plan:
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Defining Inbound Static Called Numbers and Dynamic Called Number Pool for Combined Static and Dynamic Dial Plan
With a combined static and dynamic configuration, the secondary numbers match the static inbound voice-class called-number inbound for the incoming dial-peer first. If the voice-class called-number-pool is configured under voice-port for a specific T1 or E1 controller, dynamic secondary numbers are chosen. Static secondary numbers are chosen only if no dynamic secondary number pool is found under the voice port.
In a combined static and dynamic H.320 secondary call dial plan, the inbound called number table and dynamic called number pool are configured on the same gateway.
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Use the following procedure to define a static inbound called number table and a dynamic called number pool and to assign the number pool to the voice port.
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Multiple voice ports can be configured with the same called number pool as shown in the following example.
voice class called number pool 10index 1 4085550100 - 4085550111voice-port 2/0:15voice-class called-number-pool 10voice-port 1/0:23voice-class called-number-pool 10voice class called number inbound 200index 1 40844420..Troubleshooting Tips
Use the show voice port command to verify voice port configuration.
Configuring the Outbound Static Called Numbers for Combined Static and Dynamic Dial Plan
Define the outbound stated called number table on a separate gateway. Use the following steps to define the outbound called number table.
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This example configuration shows multiple indexes defined for an outbound voice class called number.
voice class called number outbound 300index 1 4085550101index 2 4085550102index 3 4085550103index 4 4085550104index 5 4085550105index 6 4085550106index 7 4085550107Configuring Combined Static and Dynamic Dial Plan Inbound POTS Dial Peer for Originating Gateway
The same inbound dial peer is used to support both dynamic and static incoming H.320 secondary calls. The static inbound called number table is used to select a primary call when dynamic called numbers are not allocated for a primary call. Use the following steps to configure the inbound POTS dial peer for the originating gateway in a combined static and dynamic H.320 secondary call dial plan.
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dial-peer voice 12 potsincoming called-number 408information-type videovoice-class called-number inbound 200bandwidth maximum 256 minimum 64direct-inward-dialTroubleshooting Tips
Use the show dial-peer voice command to verify the dial peer configuration.
Configuring Dynamic Outbound POTS Dial Peers for Terminating Gateway
The outbound POTS dial peers on the terminating gateway handle outgoing H.320 primary and secondary calls. Configure separate dial peers for H.242 and nonH.242 endpoints.
The dynamic H.320 outbound dial peer with routing dialed numbers terminates H.242 endpoints. On the dynamic outbound POTS dial peer, called numbers are allocated from the dynamic called number pool configured on the voice port.
Use the following steps to configure an outbound POTS dial peer for a terminating gateway.
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dial-peer voice 22 potsdestination-pattern 4085550100information-type videobandwidth maximum 512no digit-striptrunkgroup isdntgTroubleshooting Tips
Use the show dial-peer voice command to verify the dial peer configuration.
Configuring Static Outbound POTS Dial Peers for Terminating Gateway
The outbound POTS dial peers on the terminating gateway handle outgoing H.320 primary and secondary calls. Configure separate dial peers for H.242 and nonH.242 endpoints.
The static outbound dial peer with routing dialed numbers terminates to nonH.242 endpoints. On the static outbound POTS dial peer, called numbers are allocated from the inbound called number table.
Use the following steps to configure the static outbound POTS dial peer on the terminating gateway.
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The following example configuration shows a static outbound POTS dial peer for a terminating gateway.
dial-peer voice 2222 potsdestination-pattern 4085550100information-type videovoice-class called-number outbound 50bandwidth maximum 256 minimum 64no digit-striptrunkgroup isdntgTroubleshooting Tips
Use the show dial-peer voice command to verify the dial peer configuration.
Configuration Examples for Integrated Data, Voice, and Video Services for ISDN Interfaces
This section provides the following configuration examples:
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Integrated Services with Combined Static and Dynamic H.320 Secondary Call Dial Plan: Example
The following example shows a combined static and dynamic H.320 secondary call dial plan. The dynamic dial plan is configured on the voice ports and the static dial plan is configured on the dial peers.
version 12.4service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router1!boot-start-markerboot-end-marker!logging buffered 4096000 debuggingno logging console!no aaa new-model!resource manager!no network-clock-participate slot 1ip subnet-zeroip cef!no ip dhcp use vrf connected!no ftp-server write-enableisdn switch-type basic-net3voice-card 1no dspfarm!voice service voiph323call start slowh245 caps mode restricted!voice class codec 1codec preference 1 g728codec preference 2 g711ulawcodec preference 3 g711alaw!voice class called number inbound 3index 1 5550100!voice class called number outbound 3index 1 5550120index 2 5550121index 3 5550122index 4 5550123!voice class called number pool 1index 1 5550130 - 5550133!interface FastEthernet0/0ip address 10.7.50.103 255.255.0.0duplex autospeed auto!interface FastEthernet0/1no ip addressshutdownduplex autospeed auto!interface BRI1/0no ip addressisdn switch-type basic-net3isdn protocol-emulate networkisdn layer1-emulate networkisdn calling-number 12345isdn supp-service name callingisdn skipsend-idverifyisdn integrate calltype all!interface BRI1/1no ip addressisdn switch-type basic-net3isdn protocol-emulate networkisdn layer1-emulate networkisdn calling-number 98765isdn skipsend-idverifyisdn integrate calltype all!interface BRI1/2no ip addressisdn switch-type basic-net3isdn protocol-emulate networkisdn layer1-emulate networkisdn calling-number 98765isdn skipsend-idverifyisdn integrate calltype all!interface BRI1/3no ip addressisdn switch-type basic-net3isdn protocol-emulate networkisdn layer1-emulate networkisdn calling-number 98765isdn skipsend-idverifyisdn integrate calltype all!ip default-gateway 10.7.0.1ip classlessip route 172.16.254.254 255.255.255.255 FastEthernet0/0!ip http server!control-plane!voice-port 1/0/0voice-class called-number-pool 1!voice-port 1/0/1voice-class called-number-pool 1!voice-port 1/1/0voice-class called-number-pool 1!voice-port 1/1/1voice-class called-number-pool 1!dial-peer voice 1 potsinformation-type videovoice-class called-number inbound 3incoming called-number 5550100bandwidth maximum 128direct-inward-dial!dial-peer voice 2 voipshutdowndestination-pattern 5550100session target ipv4:10.7.50.201codec g711ulaw!dial-peer voice 3 voipshutdowndestination-pattern 5550100voice-class codec 1session target ipv4:10.7.50.50!dial-peer voice 4 potsdestination-pattern 5550120information-type videodirect-inward-dialport 1/1/1!dial-peer voice 5 voipdestination-pattern 5550120session target ipv4:10.7.50.50!dial-peer voice 6 voipdestination-pattern 5550155session target ipv4:10.7.50.12!line con 0line aux 0line vty 0 4login!endIntegrated Services with Static H.320 Secondary Call Dial Plan: Example
The following example shows a static H.320 secondary call dial plan for calls between an SCCP endpoint and an H.320 endpoint:
version 12.4service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname Router2!boot-start-markerboot-end-marker!logging buffered 1000000 debuggingno logging console!no aaa new-model!resource manager!no network-clock-participate slot 2ip subnet-zeroip cef!no ip dhcp use vrf connected!ip dhcp pool phone1host 10.7.50.114 255.255.0.0client-identifier 0100.ffff.ffff.ffffdefault-router 10.7.50.211option 150 ip 10.7.50.211!no ip domain lookupno ftp-server write-enableisdn switch-type primary-nivoice-card 2no dspfarm!trunk group 1!voice service voiph323call start slow!voice class called number inbound 1index 1 7001!voice class called number inbound 201index 1 2001!voice class called number inbound 202index 1 2002!voice class called number inbound 203index 1 2003!voice class called number inbound 204index 1 2004!voice class called number inbound 205index 1 2005!voice class called number inbound 206index 1 2006!voice class called number inbound 207index 1 2007!voice class called number inbound 9001index 1 9001!voice class called number inbound 9999index 1 9997index 2 9998index 3 9999!voice class called number outbound 1index 1 6001!voice class called number outbound 7101index 1 7101!voice class called number outbound 1111index 1 1111index 2 1112index 3 1113index 4 1114!voice class called number pool 8888index 1 5550190index 2 5550191 - 5550199!controller T1 2/0framing esflinecode b8zspri-group timeslots 1-24!controller T1 2/1framing esflinecode b8zspri-group timeslots 1-20,24!interface FastEthernet0/0ip address 10.7.50.211 255.255.0.0duplex autospeed autoh323-gateway voip interfaceh323-gateway voip id dralion_gk ipaddr 10.7.50.49 1719h323-gateway voip h323-id b2b_3725h323-gateway voip tech-prefix 86001!interface FastEthernet0/1ip address 10.0.0.7 255.255.255.0shutdownduplex autospeed auto!interface BRI2/0no ip addressisdn switch-type basic-niisdn point-to-point-setup!interface Serial2/0:23no ip addressisdn switch-type primary-niisdn integrate calltype allno cdp enable!interface BRI2/1no ip addressisdn switch-type basic-niisdn point-to-point-setup!interface Serial2/1:23no ip addressisdn switch-type primary-niisdn integrate calltype allno cdp enable!ip default-gateway 10.7.0.1ip classlessip route 172.16.254.254 255.255.255.255 10.5.0.1ip route 172.16.254.254 255.255.255.255 FastEthernet0/0!ip http server!tftp-server flash:P00000000111.bintftp-server flash:P00000000222.bintftp-server flash:P00000000333.loadstftp-server flash:P00000000444.sbntftp-server flash:P00000000555.sb2!control-plane!voice-port 2/0:23!voice-port 2/1/0!voice-port 2/1/1!voice-port 2/1:23!dial-peer voice 3201 potsdestination-pattern 86001information-type videovoice-class called-number outbound 1bandwidth maximum 384direct-inward-dialport 2/0:23forward-digits 4!dial-peer voice 348906 voipdestination-pattern 348906video codec h263+session target ipv4:10.7.50.107req-qos controlled-load!dial-peer voice 7001 potsinformation-type videovoice-class called-number inbound 1incoming called-number 7001bandwidth maximum 384direct-inward-dial!dial-peer voice 9001 voipdestination-pattern 9001session target ipv4:10.7.50.107codec g711ulaw!dial-peer voice 2001 potsinformation-type videovoice-class called-number inbound 201incoming called-number 2001bandwidth maximum 192direct-inward-dial!dial-peer voice 2002 potsinformation-type videovoice-class called-number inbound 202incoming called-number 2002bandwidth maximum 192direct-inward-dial!dial-peer voice 2003 potsinformation-type videovoice-class called-number inbound 203incoming called-number 2003bandwidth maximum 192direct-inward-dial!dial-peer voice 2004 potsinformation-type videovoice-class called-number inbound 204incoming called-number 2004bandwidth maximum 192direct-inward-dial!dial-peer voice 2005 potsinformation-type videovoice-class called-number inbound 205incoming called-number 2005bandwidth maximum 192direct-inward-dial!dial-peer voice 2006 potsinformation-type videovoice-class called-number inbound 206incoming called-number 2006bandwidth maximum 192direct-inward-dial!dial-peer voice 7101 potsdestination-pattern 7101information-type videovoice-class called-number outbound 7101bandwidth maximum 384direct-inward-dialport 2/0:23forward-digits all!dial-peer voice 99001 potsinformation-type videovoice-class called-number inbound 9001incoming called-number 9001bandwidth maximum 384direct-inward-dial!gatewaytimer receive-rtp 1200!telephony-servicevideoload 7960-7940 P00000000111max-ephones 20max-dn 20ip source-address 10.7.50.211 port 2000service phone videoCapability 1create cnf-files version-stamp Jan 01 2002 00:00:00max-conferences 8 gain -6call-forward pattern .Ttransfer-system full-blindtransfer-pattern 6..transfer-pattern 5..transfer-pattern 4..transfer-pattern 2..transfer-pattern .Ttransfer-pattern ....!ephone-dn 1 dual-linenumber 2001application default!ephone-dn 2 dual-linenumber 2002!ephone-dn 3 dual-linenumber 2003!ephone-dn 4 dual-linenumber 2004!ephone-dn 5 dual-linenumber 2005!ephone-dn 20number 7001!ephone 1videomac-address ffff.ffff.fff1type 7960button 1:1!ephone 2videomac-address ffff.ffff.fff2type 7960button 1:2!ephone 3videomac-address ffff.ffff.fff3type 7960button 1:3!ephone 4videomac-address ffff.ffff.fff4type 7960button 1:4!ephone 5videomac-address ffff.ffff.fff5type 7960button 1:5!ephone 20videomac-address ffff.ffff.fff6type 7960button 1:20!line con 0line aux 0line vty 0 4login!endAdditional References
The following sections provide references related to integrated data, voice, and video services for ISDN interfaces.
Related Documents
Information on integrating data and voice
Integrating Data and Voice Services for ISDN PRI Interfaces on Multiservice Access Routers
ISDN configuration information
ISDN voice interface information
Video command reference information
Video telephony
Voice command reference information
Voice configuration information
Standards
MIBs
RFCs
RFC 2190
RFC 2198
RFC 2429
RTP Payload Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+)
Technical Assistance
Command Reference
This section documents the following new and modified commands:
New Commands
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Modified Commands
bandwidth (dial-peer)
To set the maximum bandwidth on a POTS dial peer for an H.320 call, use the bandwidth command in dial-peer configuration mode. To remove the bandwidth setting, use the no form of this command.
bandwidth maximum value [maximum value]
no bandwidth
Syntax Description
Command Default
No maximum bandwidth is set.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
Use this command to set the maximum and minimum bandwidth for an H.320 POTS dial-peer. Only the maximum bandwidth is required. The value must be entered in increments of 64 kbps. The minimum bandwidth setting is optional, and the value must be either 64 kbps or equal to the maximum value setting.
Examples
The following example shows configuration for POTS dial peer 200 with a maximum bandwidth of 1024 kbps:
dial-peer voice 200 potsbandwidth maximum 1024The following example shows configuration for POTS dial peer 11 with a maximum bandwidth of 640 and a minimum of 64:
dial-peer voice 11 potsbandwidth maximum 640 minimum 64Related Commands
bandwidth
Specifies the maximum aggregate bandwidth for H.323 traffic and verifies the available bandwidth of the destination gatekeeper.
debug voice h221
To debug telephony call control information, use the debug voice h221 command in privileged EXEC mode. To disable debugging output, use the no form of this command.
debug voice h221 [all | default | error [call [informational] | software [informational]] | function | individual | inout | raw [decode]]
no debug voice h221
Syntax Description
Command Modes
Privileged EXEC
Command History
Usage Guidelines
This command enables debugging for H.221 message events (voice telephony call control information).
Note
Caution
Use the debug voice h221 individual x command, (where x is an index number for a debug category), to activate a single debug, selected by index number instead of entering a group of debug commands. See Table 1 for a list of debug categories and corresponding index numbers.
Examples
The raw keyword displays the raw BAS information coming from or to the DSP. It is displayed in a hexadecimal octet format. The decode option decodes the BAS information into a readable English format.
The following is sample output from the debug voice h221 raw decode command:
BAS=81:1 0 0 0 0 0 0 1: AUDIO CAPS=g711 a-lawBAS=82:1 0 0 0 0 0 1 0: AUDIO CAPS=g711 u-lawBAS=84:1 0 0 0 0 1 0 0: AUDIO CAPS=g722 48kBAS=85:1 0 0 0 0 1 0 1: AUDIO CAPS=g728BAS=F9:1 1 1 1 1 0 0 1: H.242 MBE start indicationBAS=02:0 0 0 0 0 0 1 0: H.242 MBE length=2BAS=0A:0 0 0 0 1 0 1 0: H.242 MBE type=H.263 capsBAS=8A:1 - - - - - - -: Always 1BAS=8A:- 0 0 0 1 - - -: H.263 MPI=1BAS=8A:- - - - - 0 1 -: H.263 FORMAT=h.263_cifBAS=8A:- - - - - - - 0: No additional optionsRelated Commands
debug voip h221
To debug telephony call control information, use the debug voip h221 command in privileged EXEC mode. To disable debugging output, use the no form of this command.
debug voip h221 [all | default | error [call [informational] | software [informational]] | function | individual | inout | raw [decode]]
no debug voip h221
Syntax Description
Command Modes
Privileged EXEC
Command History
Usage Guidelines
This command enables debugging for H.221 message events (voice telephony call control information).
Note
Caution
Use the debug voip h221 individual x command, (where x is an index number for a debug category), to activate a single debug, selected by index number instead of entering a group of debug commands. See Table 2 for a list of debug categories and corresponding index numbers.
Examples
The raw keyword displays the raw BAS information coming from or to the DSP. It is displayed in a hexadecimal octet format. The decode option decodes the BAS information into a readable English format.
The following is sample output from the debug voip h221 raw decode command:
BAS=81:1 0 0 0 0 0 0 1: AUDIO CAPS=g711 a-lawBAS=82:1 0 0 0 0 0 1 0: AUDIO CAPS=g711 u-lawBAS=84:1 0 0 0 0 1 0 0: AUDIO CAPS=g722 48kBAS=85:1 0 0 0 0 1 0 1: AUDIO CAPS=g728BAS=F9:1 1 1 1 1 0 0 1: H.242 MBE start indicationBAS=02:0 0 0 0 0 0 1 0: H.242 MBE length=2BAS=0A:0 0 0 0 1 0 1 0: H.242 MBE type=H.263 capsBAS=8A:1 - - - - - - -: Always 1BAS=8A:- 0 0 0 1 - - -: H.263 MPI=1BAS=8A:- - - - - 0 1 -: H.263 FORMAT=h.263_cifBAS=8A:- - - - - - - 0: No additional optionsRelated Commands
index (voice class)
To define one or more numbers for a voice class called number, or a range of numbers for a voice class called number pool, use the index command in voice class configuration mode. To remove the number or range of numbers, use the no form of this command.
index number called-number
no index number called-number
Syntax Description
number
Digits that identify this index. Range is 1 to 2147483647.
called-number
Specifies a called number, or a range of called numbers, in E.164 format.
Command Default
No index is configured.
Command Modes
Voice class configuration
Command History
Usage Guidelines
Use this command to define one or more numbers for a voice class called number, or a range of numbers for a voice class called number pool. You can define multiple indexes for any inbound or outbound voice class called number or voice class called number pool.
When defining a range of numbers for a called number pool:
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Examples
The following example shows the configuration for indexes in voice class called number pool 100:
voice class called number pool 100index 1 4085550100 - 4085550111 (Range of called numbers are 4085550100 up to 4085550111)index 2 +3227045000The following example shows configuration for indexes in voice class called number outbound 222:
voice class called number outbound 222index 1 4085550101index 2 4085550102index 2 4085550103Related Commands
voice class called number
One or more called numbers configured for a voice class.
information-type
To select a specific information type for a Voice over IP (VoIP) or plain old telephone service (POTS) dial peer, use the information-type command in dial-peer configuration mode. To remove the current information type setting, use the no form of this command. To return to the default configuration, use the default form of this command.
information-type {fax | voice | video}
no information-type
default information-type
Syntax Description
fax
The information type is set to store-and-forward fax.
voice
The information type is set to voice. This is the default.
video
The information type is set to video.
Command Default
Voice
Command Modes
Dial peer configuration
Command History
Usage Guidelines
The fax keyword applies to both on-ramp and off-ramp store-and-forward fax functions.
Examples
The following example shows the configuration for information type fax for VoIP dial peer 10:
dial-peer voice 10 voipinformation-type faxThe following example shows the configuration for information type video for POTS dial peer 22:
dial-peer voice 22 potsinformation-type videoRelated Commands
isdn integrate calltype all
Enables integrated mode (for data, voice, and video) on ISDN BRI or PRI interfaces.
rtp payload-type
To identify the payload type of a Real-Time Transport Protocol (RTP) packet, use the rtp payload-type command in dial-peer configuration mode. To remove the RTP payload type, use the no form of this command.
rtp payload-type {cisco-cas-payload number | cisco-clear-channel number | cisco-codec-fax-ack number | cisco-codec-fax-ind number | cisco-codec-video-263+ number | cisco-codec-video-264 number | cisco-fax-relay number | cisco-pcm-switch-over-alaw number | cisco-pcm-switch-over-ulaw number | cisco-rtp-dtmf-relay number | nte number | nse number} [comfort-noise {13 | 19}]
no rtp payload-type {cisco-cas-payload | cisco-clear-channel | cisco-codec-fax-ack | cisco-codec-fax-ind | cisco-codec-video-263+ | cisco-codec-video-264 | cisco-fax-relay | cisco-pcm-switch-over-alaw | cisco-pcm-switch-over-ulaw | cisco-rtp-dtmf-relay | nte | nse }
Syntax Description
Command Default
No RTP payload type is configured.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
Use this command to identify the payload type of an RTP packet. For all payload types, the number range is 96 to 127 and the default is 101, with the exception of the video codec payload types:
•
•
For Session Initiation Protocol (SIP) calls, use this command after using the dtmf-relay command to choose the NTE method of dual-tone multifrequency (DTMF) relay.
Examples
The following command configuration identifies the RTP payload type as NTE 99:
Router(config-dial-peer)# rtp payload-type nte 99The following command configuration identifies the RTP payload type as cisco-codec-video-h264:
Router(config-dial-peer)# rtp payload-type cisco-codec-video-h264Related Commands
dtmf-relay
Specifies how an H.323 or SIP gateway relays DTMF tones between telephony interfaces and an IP network.
show call active video
To display call information for Signaling Connection Control Protocol (SCCP), Session Initiation Protocol, (SIP), and H.323 video calls in progress, use the show call active video command in user EXEC or privileged EXEC mode.
show call active video [brief | compact | echo-canceller call-id | id identifier]
Syntax Description
Command Default
No default behavior or values.
Command Modes
User EXEC
Privileged EXECCommand History
Usage Guidelines
Use this command to display the contents of the active video call table.
Examples
The following is sample output from the show call active video command:
Router # show call active videoTelephony call-legs: 7SIP call-legs: 0H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 7GENERIC:SetupTime=903690 msIndex=1PeerAddress=555556PeerSubAddress=PeerId=7001PeerIfIndex=106LogicalIfIndex=12ConnectTime=906160 msCallDuration=00:21:33 secCallState=4CallOrigin=2ChargedUnits=0InfoType=videoTransmitPackets=64654TransmitBytes=10861872ReceivePackets=129336ReceiveBytes=10346880TELE:ConnectionId=[0x9166F770 0x34D311DA 0x80080012 0x803F3110]IncomingConnectionId=[0x9166F770 0x34D311DA 0x80080012 0x803F3110]CallID=10TxDuration=0 msVoiceTxDuration=0 msFaxTxDuration=0 msCoderTypeRate=g711ulawNoiseLevel=0ACOMLevel=0OutSignalLevel=0InSignalLevel=0InfoActivity=0ERLLevel=0SessionTarget=ImgPages=0CallerName=CallerIDBlocked=FalseOriginalCallingNumber=555556OriginalCallingOctet=0x0OriginalCalledNumber=7001OriginalCalledOctet=0x80OriginalRedirectCalledNumber=OriginalRedirectCalledOctet=0xFFTranslatedCallingNumber=555556TranslatedCallingOctet=0x0TranslatedCalledNumber=7001TranslatedCalledOctet=0x80TranslatedRedirectCalledNumber=TranslatedRedirectCalledOctet=0xFFGwReceivedCalledNumber=7001GwReceivedCalledOctet3=0x80GwReceivedCallingNumber=555556GwReceivedCallingOctet3=0x0GwReceivedCallingOctet3a=0x80DSPIdentifier=2/1:1VIDEO:H320CallType=PrimaryVideoTransmitCodec=H263VideoReceiveCodec=H263VideoUsedBandwidth=384H221 STATS (AUDIO):TxPackets=129236TxDuration=1292360 msRxPackets=64604RxDuration=1291990 msBadHeaders=0PacketsLate=0PacketsEarly=1ReceiveDelay=85 msConcealmentDuration=0 msBufferOverflowDiscards=10H221 STATS (VIDEO):TxPackets=7693TxBytes=8214946PSC=6324GBSC=8401TxVideoFormat=3RxPackets=9514RxBytes=8185670VideoBytesConsumed=8117148FillBytesConsumed=40898670PSCPacketDrops=0LatePacket=0OutOfSequence=0BadHeader=0BadSSRC=0BadPayloadType=0BufferOverflow=0ControlHeaderOverflow=0FilteredDelay=250 msMinimumDelay=43 msMaximumDelay=1858 msRxVideoFormat=3GENERIC:SetupTime=903700 msIndex=1PeerAddress=7001PeerSubAddress=PeerId=20006PeerIfIndex=127LogicalIfIndex=126ConnectTime=906150 msCallDuration=00:21:35 secCallState=4CallOrigin=1ChargedUnits=0InfoType=speechTransmitPackets=0TransmitBytes=0ReceivePackets=64768ReceiveBytes=10362880TELE:ConnectionId=[0x9166F770 0x34D311DA 0x80080012 0x803F3110]IncomingConnectionId=[0x9166F770 0x34D311DA 0x80080012 0x803F3110]CallID=11TxDuration=1294180 msVoiceTxDuration=1294180 msFaxTxDuration=0 msCoderTypeRate=g711ulawNoiseLevel=0ACOMLevel=0OutSignalLevel=0InSignalLevel=0InfoActivity=2ERLLevel=0EchoCancellerMaxReflector=62709SessionTarget=ImgPages=0CallerName=CallerIDBlocked=FalseAlertTimepoint=903700 msOriginalCallingNumber=555556OriginalCallingOctet=0x0OriginalCalledNumber=7001OriginalCalledOctet=0x80OriginalRedirectCalledNumber=OriginalRedirectCalledOctet=0xFFTranslatedCallingNumber=555556TranslatedCallingOctet=0x0TranslatedCalledNumber=7001TranslatedCalledOctet=0x80TranslatedRedirectCalledNumber=TranslatedRedirectCalledOctet=0xFFGwReceivedCalledNumber=7001GwReceivedCalledOctet3=0x80GwReceivedCallingNumber=555556GwReceivedCallingOctet3=0x0GwReceivedCallingOctet3a=0x80GwOutpulsedCallingNumber=555556GwOutpulsedCallingOctet3=0x0GwOutpulsedCallingOctet3a=0x80VIDEO:H320CallType=NoneVideoTransmitCodec=NoneVideoReceiveCodec=NoneVideoCap_Codec=H263VideoCap_Format=CIFVideoUsedBandwidth=3101GENERIC:SetupTime=903910 msIndex=1PeerAddress=555556PeerSubAddress=PeerId=7001PeerIfIndex=106LogicalIfIndex=13ConnectTime=906160 msCallDuration=00:21:36 secCallState=4CallOrigin=2ChargedUnits=0InfoType=videoTransmitPackets=0TransmitBytes=0ReceivePackets=0ReceiveBytes=0TELE:ConnectionId=[0x918888CA 0x34D311DA 0x80090012 0x803F3110]IncomingConnectionId=[0x9166F770 0x34D311DA 0x80080012 0x803F3110]CallID=12TxDuration=0 msVoiceTxDuration=0 msFaxTxDuration=0 msCoderTypeRate=NoneNoiseLevel=0ACOMLevel=0OutSignalLevel=0InSignalLevel=0InfoActivity=0ERLLevel=0SessionTarget=ImgPages=0CallerName=CallerIDBlocked=FalseOriginalCallingNumber=555556OriginalCallingOctet=0x0OriginalCalledNumber=7001OriginalCalledOctet=0x80OriginalRedirectCalledNumber=OriginalRedirectCalledOctet=0xFFTranslatedCallingNumber=555556TranslatedCallingOctet=0x0TranslatedCalledNumber=7001TranslatedCalledOctet=0x80TranslatedRedirectCalledNumber=TranslatedRedirectCalledOctet=0xFFGwReceivedCalledNumber=7001GwReceivedCalledOctet3=0x80GwReceivedCallingNumber=555556GwReceivedCallingOctet3=0x0GwReceivedCallingOctet3a=0x80VIDEO:H320CallType=SecondaryGENERIC:SetupTime=904230 msIndex=1PeerAddress=555556PeerSubAddress=PeerId=7001PeerIfIndex=106LogicalIfIndex=14ConnectTime=906160 msCallDuration=00:21:37 secCallState=4CallOrigin=2ChargedUnits=0InfoType=videoTransmitPackets=0TransmitBytes=0ReceivePackets=0ReceiveBytes=0TELE:ConnectionId=[0x91B95C6E 0x34D311DA 0x800A0012 0x803F3110]IncomingConnectionId=[0x9166F770 0x34D311DA 0x80080012 0x803F3110]CallID=13TxDuration=0 msVoiceTxDuration=0 msFaxTxDuration=0 msCoderTypeRate=NoneNoiseLevel=0ACOMLevel=0OutSignalLevel=0InSignalLevel=0InfoActivity=0ERLLevel=0SessionTarget=ImgPages=0CallerName=CallerIDBlocked=FalseOriginalCallingNumber=555556OriginalCallingOctet=0x0OriginalCalledNumber=7001OriginalCalledOctet=0x80OriginalRedirectCalledNumber=OriginalRedirectCalledOctet=0xFFTranslatedCallingNumber=555556TranslatedCallingOctet=0x0TranslatedCalledNumber=7001TranslatedCalledOctet=0x80TranslatedRedirectCalledNumber=TranslatedRedirectCalledOctet=0xFFGwReceivedCalledNumber=7001GwReceivedCalledOctet3=0x80GwReceivedCallingNumber=555556GwReceivedCallingOctet3=0x0GwReceivedCallingOctet3a=0x80VIDEO:H320CallType=SecondaryGENERIC:SetupTime=904550 msIndex=1PeerAddress=555556PeerSubAddress=PeerId=7001PeerIfIndex=106LogicalIfIndex=15ConnectTime=906160 msCallDuration=00:21:40 secCallState=4CallOrigin=2ChargedUnits=0InfoType=videoTransmitPackets=0TransmitBytes=0ReceivePackets=0ReceiveBytes=0TELE:ConnectionId=[0x91EA317E 0x34D311DA 0x800B0012 0x803F3110]IncomingConnectionId=[0x9166F770 0x34D311DA 0x80080012 0x803F3110]CallID=14TxDuration=0 msVoiceTxDuration=0 msFaxTxDuration=0 msCoderTypeRate=NoneNoiseLevel=0ACOMLevel=0OutSignalLevel=0InSignalLevel=0InfoActivity=0ERLLevel=0SessionTarget=ImgPages=0CallerName=CallerIDBlocked=FalseOriginalCallingNumber=555556OriginalCallingOctet=0x0OriginalCalledNumber=7001OriginalCalledOctet=0x80OriginalRedirectCalledNumber=OriginalRedirectCalledOctet=0xFFTranslatedCallingNumber=555556TranslatedCallingOctet=0x0TranslatedCalledNumber=7001TranslatedCalledOctet=0x80TranslatedRedirectCalledNumber=TranslatedRedirectCalledOctet=0xFFGwReceivedCalledNumber=7001GwReceivedCalledOctet3=0x80GwReceivedCallingNumber=555556GwReceivedCallingOctet3=0x0GwReceivedCallingOctet3a=0x80VIDEO:H320CallType=SecondaryGENERIC:SetupTime=904870 msIndex=1PeerAddress=555556PeerSubAddress=PeerId=7001PeerIfIndex=106LogicalIfIndex=16ConnectTime=906160 msCallDuration=00:21:41 secCallState=4CallOrigin=2ChargedUnits=0InfoType=videoTransmitPackets=0TransmitBytes=0ReceivePackets=0ReceiveBytes=0TELE:ConnectionId=[0x921B0522 0x34D311DA 0x800C0012 0x803F3110]IncomingConnectionId=[0x9166F770 0x34D311DA 0x80080012 0x803F3110]CallID=15TxDuration=0 msVoiceTxDuration=0 msFaxTxDuration=0 msCoderTypeRate=NoneNoiseLevel=0ACOMLevel=0OutSignalLevel=0InSignalLevel=0InfoActivity=0ERLLevel=0SessionTarget=ImgPages=0CallerName=CallerIDBlocked=FalseOriginalCallingNumber=555556OriginalCallingOctet=0x0OriginalCalledNumber=7001OriginalCalledOctet=0x80OriginalRedirectCalledNumber=OriginalRedirectCalledOctet=0xFFTranslatedCallingNumber=555556TranslatedCallingOctet=0x0TranslatedCalledNumber=7001TranslatedCalledOctet=0x80TranslatedRedirectCalledNumber=TranslatedRedirectCalledOctet=0xFFGwReceivedCalledNumber=7001GwReceivedCalledOctet3=0x80GwReceivedCallingNumber=555556GwReceivedCallingOctet3=0x0GwReceivedCallingOctet3a=0x80VIDEO:H320CallType=SecondaryGENERIC:SetupTime=905190 msIndex=1PeerAddress=555556PeerSubAddress=PeerId=7001PeerIfIndex=106LogicalIfIndex=17ConnectTime=906160 msCallDuration=00:21:42 secCallState=4CallOrigin=2ChargedUnits=0InfoType=videoTransmitPackets=0TransmitBytes=0ReceivePackets=0ReceiveBytes=0TELE:ConnectionId=[0x924BD82D 0x34D311DA 0x800D0012 0x803F3110]IncomingConnectionId=[0x9166F770 0x34D311DA 0x80080012 0x803F3110]CallID=16TxDuration=0 msVoiceTxDuration=0 msFaxTxDuration=0 msCoderTypeRate=NoneNoiseLevel=0ACOMLevel=0OutSignalLevel=0InSignalLevel=0InfoActivity=0ERLLevel=0SessionTarget=ImgPages=0CallerName=CallerIDBlocked=FalseOriginalCallingNumber=555556OriginalCallingOctet=0x0OriginalCalledNumber=7001OriginalCalledOctet=0x80OriginalRedirectCalledNumber=OriginalRedirectCalledOctet=0xFFTranslatedCallingNumber=555556TranslatedCallingOctet=0x0TranslatedCalledNumber=7001TranslatedCalledOctet=0x80TranslatedRedirectCalledNumber=TranslatedRedirectCalledOctet=0xFFGwReceivedCalledNumber=7001GwReceivedCalledOctet3=0x80GwReceivedCallingNumber=555556GwReceivedCallingOctet3=0x0GwReceivedCallingOctet3a=0x80VIDEO:H320CallType=SecondaryTelephony call-legs: 7SIP call-legs: 0H323 call-legs: 0Call agent controlled call-legs: 0SCCP call-legs: 0Multicast call-legs: 0Total call-legs: 7Table 3 describes significant fields shown in this output.
Related Commands
show dial-peer voice
To display information for voice dial peers, use the show dial-peer voice command in user EXEC or privileged EXEC mode.
show dial-peer voice [number | summary]
Syntax Description
number
(Optional) A specific voice dial peer. Output displays detailed information about that dial peer.
summary
(Optional) Output displays a short summary of each voice dial peer.
Command Default
If both the name argument and summary keyword are omitted, output displays detailed information about all voice dial peers.
Command Modes
User EXEC
Privileged EXECCommand History
Usage Guidelines
Use this command to display the configuration for all VoIP and POTS dial peers configured for a gateway. To show configuration information for only one specific dial peer, use the number argument to identify the dial peer. To show summary information for all dial peers, use the summary keyword.
Examples
The following is sample output from the show dial-peer voice command for a POTS dial peer:
Router# show dial-peer voice 100VoiceEncapPeer3201peer type = voice, information type = video,description = `',tag = 3201, destination-pattern = `86001',answer-address = `', preference=0,CLID Restriction = NoneCLID Network Number = `'CLID Second Number sentCLID Override RDNIS = disabled,source carrier-id = `', target carrier-id = `',source trunk-group-label = `', target trunk-group-label = `',numbering Type = `unknown'group = 3201, Admin state is up, Operation state is up,Outbound state is up,incoming called-number = `', connections/maximum = 0/unlimited,DTMF Relay = disabled,URI classes:Destination =huntstop = disabled,in bound application associated: 'DEFAULT'out bound application associated: ''dnis-map =permission :bothincoming COR list:maximum capabilityoutgoing COR list:minimum requirementTranslation profile (Incoming):Translation profile (Outgoing):incoming call blocking:translation-profile = `'disconnect-cause = `no-service'advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4type = pots, prefix = `',forward-digits 4session-target = `', voice-port = `2/0:23',direct-inward-dial = enabled,digit_strip = enabled,register E.164 number with H323 GK and/or SIP Registrar = TRUEfax rate = system, payload size = 20 bytessupported-language = ''preemption level = `routine'bandwidth:maximum = 384 KBits/sec, minimum = 64 KBits/secvoice class called-number:inbound = `', outbound = `1'Time elapsed since last clearing of voice call statistics neverConnect Time = 0, Charged Units = 0,Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0Accepted Calls = 0, Refused Calls = 0,Last Disconnect Cause is "",Last Disconnect Text is "",Last Setup Time = 0.The following is sample output from this command for a VoIP dial peer:
Router# show dial-peer voice 101VoiceOverIpPeer101peer type = voice, information type = voice,description = `',tag = 6001, destination-pattern = `6001',answer-address = `', preference=0,CLID Restriction = NoneCLID Network Number = `'CLID Second Number sentCLID Override RDNIS = disabled,source carrier-id = `', target carrier-id = `',source trunk-group-label = `', target trunk-group-label = `',numbering Type = `unknown'group = 6001, Admin state is up, Operation state is up,incoming called-number = `', connections/maximum = 0/unlimited,DTMF Relay = disabled,modem transport = system,URI classes:Incoming (Called) =Incoming (Calling) =Destination =huntstop = disabled,in bound application associated: 'DEFAULT'out bound application associated: ''dnis-map =permission :bothincoming COR list:maximum capabilityoutgoing COR list:minimum requirementTranslation profile (Incoming):Translation profile (Outgoing):incoming call blocking:translation-profile = `'disconnect-cause = `no-service'advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4type = voip, session-target = `ipv4:1.7.50.50',technology prefix:settle-call = disabledip media DSCP = ef, ip signaling DSCP = af31,ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41ip video rsvp-fail DSCP = af41,UDP checksum = disabled,session-protocol = cisco, session-transport = system,req-qos = best-effort, acc-qos = best-effort,req-qos video = best-effort, acc-qos video = best-effort,req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,RTP dynamic payload type values: NTE = 101Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122CAS=123, ClearChan=125, PCM switch over u-law=0,A-law=8h263+=118, h264=119RTP comfort noise payload type = 19fax rate = fax, payload size = 20 bytesfax protocol = systemfax-relay ecm enablefax NSF = 0xAD0051 (default)codec = g711ulaw, payload size = 160 bytes,video codec = h263+voice class codec = `'Media Setting = flow-through (global)Expect factor = 10, Icpif = 20,Playout Mode is set to adaptive,Initial 60 ms, Max 250 msPlayout-delay Minimum mode is set to default, value 40 msFax nominal 300 msMax Redirects = 1, signaling-type = cas,VAD = enabled, Poor QOV Trap = disabled,Source Interface = NONEvoice class sip url = system,voice class sip rel1xx = system,redirect ip2ip = disabledprobe disabled,voice class perm tag = `'Time elapsed since last clearing of voice call statistics neverConnect Time = 0, Charged Units = 0,Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0Accepted Calls = 0, Refused Calls = 0,Last Disconnect Cause is "",Last Disconnect Text is "",Last Setup Time = 0.Table 4 describes significant fields shown in this output.
The following is sample output from this command with the summary keyword:
Router# show dial-peer voice summarydial-peer hunt 0PASSTAG TYPE ADMIN OPER PREFIX DEST-PATTERN PREF THRU SESS-TARGET PORT100 pots up up 0101 voip up up 5550112 0 syst ipv4:10.10.1.1102 voip up up 5550134 0 syst ipv4:10.10.1.199 voip up down 0 syst33 pots up down 0Table 5 describes significant fields shown in this output.
Related Commands
show voice class called-number
To display a specific voice class called-number, use the show voice class called-number command in privileged EXEC mode.
show voice class called-number [inbound | outbound] tag
Syntax Description
inbound
Displays the specified inbound voice class called-number.
outbound
Displays the specified outbound voice class called-number.
tag
Digits that identify this voice class called-number.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use this command to display a specific inbound or outbound voice class called-number.
Examples
The following is sample output from this command:
Router# show voice class called-number outbound 200Called Number Outbound: 200index 1 4085550100index 2 4085550102index 3 4085550103index 4 4085550104Table 6 describes significant fields shown in the display.
Related Commands
show voice class called-number-pool
Displays voice class called number pool configuration information.
show voice class called-number-pool
To display a voice class called-number pool, use the show voice class called-number-pool command in privileged EXEC mode.
show voice class called-number-pool tag [detail]
Syntax Description
tag
Digits that identify this voice class called-number-pool. Range is 1 to 10000.
detail
Displays idle called number and allocated called number information.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use this command to display the voice class called number pool configuration information. The detail keyword displays up to 16 idle called numbers, and up to 4 allocated called numbers for each allocated request.
Examples
The following sample output displays configuration information for voice class called-number-pool 100, including idle called numbers and allocated called numbers:
Router(config)# show voice class called-number-pool 100 detailCalled Number Pool: 100index 1 100A11 - 100A20index 2 200#55 - 200#77index 3 5551111 - 6662333index 99 123C11 - 123C99All called numbers are generated from table: FALSENo of idle called numbers: 16List of idle called numbers:100A11 100A12 .. Display up to 16 idle called number from the pool100A13 100A14100A15 100A16100A17 100A18100A19 100A20200#55 200#56200#57 200#58200#59 200#60No of alloc requests : 1Ref Id Alloc PC Size2 41F84190 16List of alloc called numbers: .. Display the first 4 allocated called number for RefId 2200#61 200#62200#63 200#64Table 7 describes significant fields shown in the display.
Related Commands
show voice dsp
To show the current status of all digital signal processor (DSP) voice channels, use the show voice dsp command in privileged EXEC mode.
show voice dsp
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use this command if abnormal behavior occurs in the DSP voice channels.
Examples
The following sample output shows the current status of the codec, set for modem relay, on channel 1.
Router# show voice dsp----------------------------FLEX VOICE CARD 1 ------------------------------*DSP VOICE CHANNELS*DSP DSP DSPWARE CURR BOOT PAK TX/RXTYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT===== === == ======== ======= ===== ======= === == ========= == ==== ============C5510 001 01 modem-re 4.5.909 busy idle 0 0 1/1/0 05 0 298/353*DSP SIGNALING CHANNELS*DSP DSP DSPWARE CURR BOOT PAK TX/RXTYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT===== === == ======== ======= ===== ======= === == ========= == ==== ============C5510 001 05 {flex} 4.5.909 alloc idle 0 0 1/1/3 02 0 15/0C5510 001 06 {flex} 4.5.909 alloc idle 0 0 1/1/2 02 0 17/0C5510 001 07 {flex} 4.5.909 alloc idle 0 0 1/1/1 06 0 31/0C5510 001 08 {flex} 4.5.909 alloc idle 0 0 1/1/0 06 0 321/0------------------------END OF FLEX VOICE CARD 1 ----------------------------The following sample output shows the current status of all DSP voice channels:
Router# show voice dspDSP# 0, channel# 0 G729A BUSYDSP# 0, channel# 1 G729A BUSYDSP# 1, channel# 2 FAX IDLEDSP# 1, channel# 3 FAX IDLEDSP# 2, channel# 4 NONE BADDSP# 2, channel# 5 NONE BADDSP# 3, channel# 6 NONE BADDSP# 3, channel# 7 NONE BADDSP# 4, channel# 8 NONE BADDSP# 4, channel# 9 NONE BADDSP# 5, channel# 10 NONE BADDSP# 5, channel# 11 NONE BADThe following is sample output from this command on a Cisco 1750 router:
Router# show voice dspDSP#0: state IN SERVICE, 2 channels allocatedchannel#0: voice port 1/0, codec G711 ulaw, state UPchannel#1: voice port 1/1, codec G711 ulaw, state UPDSP#1: state IN SERVICE, 2 channels allocatedchannel#0: voice port 2/0, codec G711 ulaw, state UPchannel#1: voice port 2/1, codec G711 ulaw, state UPDSP#2: state RESET, 0 channels allocatedThe following is sample output from this command on a secure Cisco Survivable Remote Site Telephony (Cisco SRST) router with the NM-HDV network module and the TI-549 (C549) DSP installed:
Router# show voice dspDSP DSP DSPWARE CURR BOOT PAK TX/RXTYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT==== === == ======== ======= ===== ======= === == ======== === ==== ===========C549 1 01 {medium} 4.4.3 IDLE idle 0 0 1/0:0 1 0 9357/9775C549 1 02 {medium} 4.4.3 IDLE idle 0 1/0:0 2 0 0/0C549 2 01 {medium} 4.4.3 IDLE idle 0 0 1/0:0 3 0 0/0C549 2 02 {medium} 4.4.3 IDLE idle 0 1/0:0 4 0 0/0C549 3 01 {medium} 4.4.3 IDLE idle 0 0 1/0:0 5 0 0/13C549 3 02 {medium} 4.4.3 IDLE idle 0 1/0:0 6 0 0/13The following is sample output from this command for an H.320 network configured for video support:
Router# show voice dspDSP DSP DSPWARE CURR BOOT PAK TX/RXTYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABORT PACK COUNT==== === == ======== ======= ===== ======= === == ========= == ===== ============ edsp 001 01 g711ulaw 0.1 IDLE 50/0/1.1 edsp 002 02 g711ulaw 0.1 IDLE 50/0/1.2 edsp 003 01 g729r8 p 0.1 IDLE 50/0/2.1 ----------------------------FLEX VOICE CARD 1 ------------------------------*DSP VOICE CHANNELS*DSP DSP DSPWARE CURR BOOT PAK TX/RXTYPE NUM CH CODEC VERSION STATE STATE RST AI VOICEPORT TS ABRT PACK COUNT===== === == ======== ======= ===== ======= === == ========= == ==== ============C5510 001 05 None 9.0.105 idle idle 0 0 0 0/0C5510 001 06 None 9.0.105 idle idle 0 0 0 0/0C5510 001 07 None 9.0.105 idle idle 0 0 0 0/0C5510 001 08 None 9.0.105 idle idle 0 0 0 0/0C5510 001 09 None 9.0.105 idle idle 0 0 0 0/0C5510 001 10 None 9.0.105 idle idle 0 0 0 0/0C5510 001 11 None 9.0.105 idle idle 0 0 0 0/0C5510 001 12 None 9.0.105 idle idle 0 0 0 0/0C5510 001 13 None 9.0.105 idle idle 0 0 0 0/0C5510 001 14 None 9.0.105 idle idle 0 0 0 0/0C5510 001 15 None 9.0.105 idle idle 0 0 0 0/0C5510 001 16 None 9.0.105 idle idle 0 0 0 0/0C5510 003 01 None 9.0.105 idle idle 0 0 0 0/0C5510 003 02 None 9.0.105 idle idle 0 0 0 0/0C5510 003 03 None 9.0.105 idle idle 0 0 0 0/0C5510 003 04 None 9.0.105 idle idle 0 0 0 0/0C5510 003 05 None 9.0.105 idle idle 0 0 0 0/0C5510 003 06 None 9.0.105 idle idle 0 0 0 0/0C5510 003 07 None 9.0.105 idle idle 0 0 0 0/0C5510 003 08 None 9.0.105 idle idle 0 0 0 0/0C5510 003 09 None 9.0.105 idle idle 0 0 0 0/0C5510 003 10 None 9.0.105 idle idle 0 0 0 0/0C5510 003 11 None 9.0.105 idle idle 0 0 0 0/0C5510 003 12 None 9.0.105 idle idle 0 0 0 0/0C5510 003 13 None 9.0.105 idle idle 0 0 0 0/0C5510 003 14 None 9.0.105 idle idle 0 0 0 0/0C5510 003 15 None 9.0.105 idle idle 0 0 0 0/0C5510 003 16 None 9.0.105 idle idle 0 0 0 0/0*DSP H.320 CHANNELS*DSP DSP TX/RX DSPWARE CURR PAK TX/RXTYPE NUM CH CODEC VERSION STATE VOICEPORT TS ABRT PACK COUNT===== === === =========== ======= ===== ========= == ==== ============C5510 001 01 h320p(01) 9.0.105 busy 1/0/0:15 06001 02 h320s(02) 9.0.105 busy 1/0/0:15 07001 03 h320s(03) 9.0.105 busy 1/0/0:15 08001 04 h320s(04) 9.0.105 busy 1/0/0:15 09001 01a g711ulaw 9.0.105 busy 0 1013663/508300001 01v h263 /h263 9.0.105 busy 0 104908/309114------------------------END OF FLEX VOICE CARD 1 ----------------------------Table 8 describes significant fields shown in the output.
Related Commands
show voice port
To display configuration information about a specific voice port, use the show voice port command in privileged EXEC mode.
Cisco 1750 Router
show voice port slot/port
Cisco 2600 and Cisco 3600 Series Router with Analog Voice Ports
show voice port [slot/subunit/port | summary]
Cisco 2600 and Cisco 3600 Series Router with Digital Voice Ports (with T1 Packet Voice Trunk Network Modules)
show voice port [slot/port:ds0-group | summary]
Cisco AS5300 Universal Access Server
show voice port controller-number:D
Cisco 7200 Series Router
show voice port {slot/port:ds0-group-no} | {slot/subunit/port}
Syntax Description
Cisco 1750 Router
slot
Slot number in the router in which the voice interface card (VIC) is installed. Range is 0 to 2, depending on the slot in which it is installed.
port
Voice port. Valid entries are 0 and 1.
Cisco 2600 and Cisco 3600 Series Router with Analog Voice Ports
Cisco 2600 and Cisco 3600 Series Router with Digital Voice Ports
Cisco AS5300 Access Server
Cisco 7200 Series Router
Command Modes
Privileged EXEC
Command History
Usage Guidelines
Use this command to display configuration and voice-interface-card-specific information about a specific port.
This command applies to Voice over IP, Voice over Frame Relay, and Voice over ATM.
The ds0-group command automatically creates a logical voice port that is numbered as follows on Cisco 2600, Cisco 3600 series, and Cisco 7200 series routers: slot/port:ds0-group-no. Although only one voice port is created for each group, applicable calls are routed to any channel in the group.
Examples
The following is sample output from the show voice port command for an E&M analog voice port:
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 a foreign exchange station (FXS) analog voice port:
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 E&M digital voice port:
Router# show voice port 1/0/1receEive and transMit Slot is 1, Sub-unit is 0, Port is 1Type of VoicePort is E&MOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is not setNoise 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 8 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 USThe following is sample output from the show voice port command:
Router# show voice port 1/0/1receEive and transMit Slot is 1, Sub-unit is 0, Port is 1Type of VoicePort is E&MOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is not setNoise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 DBMSIn Gain is Set to 0 dBmOut 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 sRegion Tone is set for USThe following is sample output from the show voice port command for an ISDN voice port:
Router# show voice portISDN 2/0:23 Slot is 2, Sub-unit is 0, Port is 23Type of VoicePort is ISDN-VOICEOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is not setNoise Regeneration is enabledNon Linear Processing is enabledNon Linear Mute is disabledNon Linear Threshold is -21 dBMusic On Hold Threshold is Set to -38 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancellation NLP mute is disabledEcho Cancellation NLP threshold is -21 dBEcho Cancel Coverage is set to 64 msEcho Cancel worst case ERL is set to 6 dBPlayout-delay Mode is set to adaptivePlayout-delay Nominal is set to 60 msPlayout-delay Maximum is set to 250 msPlayout-delay Minimum mode is set to default, value 40 msPlayout-delay Fax is set to 300 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sCall Disconnect Time Out is set to 60 sRinging Time Out is set to 180 sWait Release Time Out is set to 30 sCompanding Type is u-lawRegion Tone is set for USStation name None, Station number NoneTranslation profile (Incoming):Translation profile (Outgoing):Voice class called number pool:DS0 channel specific status info:IN OUTPORT CH SIG-TYPE OPER STATUS STATUS TIP RING2/0:23 01 isdn-voice up none none2/0:23 02 isdn-voice up none none2/0:23 03 isdn-voice up none none2/0:23 04 isdn-voice up none none2/0:23 05 isdn-voice up none none2/0:23 06 isdn-voice up none none2/0:23 07 isdn-voice dorm none none2/0:23 08 isdn-voice dorm none none2/0:23 09 isdn-voice dorm none none2/0:23 10 isdn-voice dorm none none2/0:23 11 isdn-voice dorm none none2/0:23 12 isdn-voice dorm none none2/0:23 13 isdn-voice dorm none none2/0:23 14 isdn-voice dorm none none2/0:23 15 isdn-voice dorm none none2/0:23 16 isdn-voice dorm none none2/0:23 17 isdn-voice dorm none none2/0:23 18 isdn-voice dorm none none2/0:23 19 isdn-voice dorm none none2/0:23 20 isdn-voice dorm none none2/0:23 21 isdn-voice dorm none none2/0:23 22 isdn-voice dorm none none2/0:23 23 isdn-voice dorm none noneTable 9 describes significant fields shown in each these output.
video codec (dial-peer)
To assign a video codec to a VoIP dial peer, use the video codec command in dial-peer configuration mode. To remove a video codec, use the no form of this command.
video codec {h261 | h263 | h263+ | h264}
no video codec
Syntax Description
Command Default
No video codec is configured.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
Use this command to configure a video codec for a VoIP dial peer. If no video codec is configured, the default is transparent codec operation between the endpoints.
Examples
The following example shows configuration for video codec H.263+ on VoIP dial peer 30:
dial-peer voice 30 voipvideo codec h263+Related Commands
video codec (voice-class)
To specify a video codec for a voice class, use the video codec command in voice class configuration mode. To remove the video codec, use the no form of this command.
video codec {h261 | h263 | h263+ | h264}
no video codec {h261 | h263 | h263+ | h264}
Syntax Description
h261
Apply this preference to video codec H.261
h263
Apply this preference to video codec H.263
h263+
Apply this preference to video codec H.263+
h264
Apply this preference to video codec H.264
Command Default
No video codec is configured.
Command Modes
Voice class configuration
Command History
Usage Guidelines
Use this command to specify one or more video codecs for a voice class.
Examples
The following example shows configuration for voice class codec 10 with two audio codec preferences and three video codec preferences:
voice class codec 10codec preference 1 g711alawcodec preference 2 g722video codec h261video codec h263video codec h264Related Commands
voice class called number
To define a voice class called number or range of numbers, use the voice class called number command in global configuration mode. To remove a voice class called number, use the no form of this command.
voice class called number {inbound | outbound | pool} tag
no voice class called number
Syntax Description
Command Default
No voice class called number is configured.
Command Modes
Global configuration
Command History
Usage Guidelines
Use this command to define one or more static voice class called numbers for inbound and outbound POTS dial peers or a dynamic voice class called number pool. The indexes for a voice class called number are defined with the index (voice class) command.
To configure the gateway to use the same called number as both primary and secondary numbers for an H.320 call, configure an outbound called-number voice-class with no index defined and apply it to the outbound POTS dial-peer as follows:
voice class called-number outbound 1dial-peer voice 1 potsvoice-class called-number outbound 1
Note
Examples
The following example shows configuration for an outbound voice class called number:
voice class called number outbound 30index 1 5550100index 2 5550101index 3 5550102index 4 5550103The following example shows configuration for a voice class called number pool:
voice class called number pool 1index 1 5550100 - 5550199Related Commands
voice-class called-number (dial peer)
To assign a previously defined voice class called number to an inbound or outbound POTS dial peer, use the voice-class called-number command in dial peer configuration mode. To remove a voice class called number from the dial peer, use the no form of this command.
voice-class called-number [inbound | outbound] tag
no voice-class called-number
Syntax Description
Command Default
No voice class called number is configured on the dial peer.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
Use this command to assign a previously defined voice class called number to a dial peer for a static H.320 secondary call dial plan. Use the inbound keyword for inbound POTS dial peers, and the outbound keyword for outbound POTS dial peers.
Note
Examples
The following example shows configuration for an outbound voice class called number outbound on POTS dial peer 22:
dial-peer voice 22 potsvoice-class called-number inbound 300Related Commands
voice-class called-number-pool
To assign a previously defined voice class called number pool to a voice port, use the voice-class called-number-pool command in voice port configuration mode. To remove a voice class called number pool from the voice port, use the no form of this command.
voice-class called-number-pool tag
no voice-class called-number-pool
Syntax Description
Command Default
No voice class called number pool is assigned to the voice port.
Command Modes
Voice class configuration
Command History
Usage Guidelines
Use this command to assign a voice class called number pool to a voice port for a dynamic H.320 secondary call dial plan.
Examples
The following example shows configuration for voice class called number pool 100 on voice port 1/0/0:
voice-port 1/0/0voice-class called-number-pool 100Related Commands
Feature Information for Integrating Data, Voice, and Video for ISDN Interfaces
Table 10 lists the release history for this feature.
Not all commands may be available in your Cisco IOS software release. For release information about a specific command, see the command reference documentation.
Use Cisco Feature Navigator to find information about platform support and software image support. Cisco Feature Navigator enables you to determine which Cisco IOS and Catalyst OS software images support a specific software release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on Cisco.com is not required.
Note
Table 10 Feature Information for Integrating Data, Voice, and Video Services for ISDN Interfaces
Cisco IOS H.320 Video Gateway
12.4(11)T
The Cisco IOS H.320 Video Gateway provides the capability to send H.320 encapsulated Audio/Video calls over TDM voice interfaces.
The following sections provide information about this feature:
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