Cisco Cisco Customer Voice Portal 8.0(1) Design Guide
4-22
Cisco Unified Customer Voice Portal (CVP) 8.x Solution Reference Network Design (SRND)
OL-15989-06
Chapter 4 Designing Unified CVP for High Availability
Unified CVP IVR Service
gateways with a list of application server IP addresses and/or using the Content Services Switch (CSS).
With Unified CVP 4.0 and later releases, the IVR Service is tightly coupled with the SIP Service or
H.323 Service. If the IVR Service goes out of service, the H.323 or SIP Service will be taken out of
service as well so that no further calls are accepted by the Unified CVP Call Server.
With Unified CVP 4.0 and later releases, the IVR Service is tightly coupled with the SIP Service or
H.323 Service. If the IVR Service goes out of service, the H.323 or SIP Service will be taken out of
service as well so that no further calls are accepted by the Unified CVP Call Server.
Configuration
No additional configuration is needed in order to tell the H.323 or SIP Service which IVR Service to use.
By default, the H.323 and SIP Service use the IVR Service that resides on the same server. It is also no
longer necessary to configure the VoiceXML gateway with the IP address of the Call Server’s IVR
Service. When SIP is used, the SIP Service inserts the URL of the Call Server's IVR Service into a header
in the SIP INVITE message when the call is sent to the VoiceXML gateway. The VoiceXML gateway
extracts this information from the SIP INVITE and uses it when determining which Call Server to use.
When H.323 is used, the VoiceXML gateway examines the source IP address of the incoming call from
the Call Server. This IP address is then used as the address for the Call Server’s IVR Service.
By default, the H.323 and SIP Service use the IVR Service that resides on the same server. It is also no
longer necessary to configure the VoiceXML gateway with the IP address of the Call Server’s IVR
Service. When SIP is used, the SIP Service inserts the URL of the Call Server's IVR Service into a header
in the SIP INVITE message when the call is sent to the VoiceXML gateway. The VoiceXML gateway
extracts this information from the SIP INVITE and uses it when determining which Call Server to use.
When H.323 is used, the VoiceXML gateway examines the source IP address of the incoming call from
the Call Server. This IP address is then used as the address for the Call Server’s IVR Service.
The following example illustrates the VoiceXML bootstrap service that is invoked when a call is
received:
received:
service bootstrap flash:bootstrap.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
Unlike Unified CVP 3.1 and earlier releases, with Unified CVP 4.0 and later releases you do not have to
configure the IP address of the Call Server. The bootstrap.tcl learns the IP address of the source Call
Server and uses it as its call server. There is no need for a CSS or backup Call Server configuration
because receiving a call from the Call Server means that the server is up and operational.
configure the IP address of the Call Server. The bootstrap.tcl learns the IP address of the source Call
Server and uses it as its call server. There is no need for a CSS or backup Call Server configuration
because receiving a call from the Call Server means that the server is up and operational.
The following files in flash memory on the gateway are also involved with high availability: handoff.tcl,
survivability.tcl, recovery.vxml, and several .wav files. Use Trivial File Transfer Protocol (TFTP) to load
the proper files into flash. Configuration information for each file can be found within the file itself. For
more information, refer to the latest version of the Configuration and Administration Guide for Cisco
Unified Customer Voice Portal (CVP), available at
survivability.tcl, recovery.vxml, and several .wav files. Use Trivial File Transfer Protocol (TFTP) to load
the proper files into flash. Configuration information for each file can be found within the file itself. For
more information, refer to the latest version of the Configuration and Administration Guide for Cisco
Unified Customer Voice Portal (CVP), available at
Call Disposition
If the Unified CVP IVR Service fails, the following conditions apply to the call disposition:
•
Calls in progress are default-routed to an alternate location by survivability on the originating
gateway. (Survivability does not apply in NIC-routing models.)
gateway. (Survivability does not apply in NIC-routing models.)
•
New calls are directed to an in-service Unified CVP IVR Service.