Step 1 |
Install the
IOS image on the Ingress Gateway.
|
Step 2 |
Transfer the
following script, configuration, and .wav files to the Ingress gateway through
the Operations Console or the Unified CVP product CD:
-
bootstrap.tcl
-
handoff.tcl
-
survivabilty.tcl
-
bootstrap.vxml
-
recovery.vxml
-
ringtone.tcl
-
cvperror.tcl
-
ringback.wav
-
critical_error.wav
|
Step 3 |
Configure the
Ingress Gateway base settings.
|
Step 4 |
Configure the
Ingress Gateway service settings.
|
Step 5 |
Configure an
Ingress Gateway incoming Pots Dial-peer.
|
Step 6 |
For
SIP without a
Proxy Server
, complete the following steps:
-
If you
are using DNS query with SRV or A types from the gateway, configure the gateway
to use DNS.
Also, if
you are using DNS query with SRV or A types from the gateway, use CLI as shown
below:
Note
|
Generally, a non-DNS setup is:
sip-server ipv4:xx.xx.xxx.xxx:5060 .
|
ip domain name pats.cisco.com
ip name-server 10.86.129.16
sip-ua
sip-server dns:cvp.pats.cisco.com
OR:
ipv4:xx.xx.xxx.xxx:5060
-
Configure
the DNS zone file for the separate DNS server that displays how the Service
(SRV) records are configured.
|
Step 7 |
For
SIP with a
Proxy Server, if you are using the DNS Server, you can set your SIP Service
as the Host Name (either A or SRV type).
You can also
configure the Gateway statically instead of using DNS. The following example
shows how both the A and SRV type records could be configured:
ip host cvp4cc2.cisco.com 10.4.33.132
ip host cvp4cc3.cisco.com 10.4.33.133
ip host cvp4cc1.cisco.com 10.4.33.131
For SIP/TCP:
ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 cvp4cc3.cisco.com
ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 cvp4cc2.cisco.com
ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 cvp4cc1.cisco.com
For SIP/UDP:
ip host _sip._udp.cvp.cisco.com srv 50 50 5060 cvp4cc3.cisco.com
ip host _sip._udp.cvp.cisco.com srv 50 50 5060 cvp4cc2.cisco.com
ip host _sip._udp.cvp.cisco.com srv 50 50 5060 cvp4cc1.cisco.com
Note
|
The DNS
Server must be configured with all necessary A type or SRV type records.
|
|
Step 8 |
Transfer
files to the
VXML
Gateway using Step 2.
|
Step 9 |
Configure the
VXML Gateway base settings.
|
Step 10 |
Configure the
VXML Gateway service settings.
|
Step 11 |
If using ASR
and TTS Servers, specify IP addresses for those servers for each locale using
the applicable name resolution system for the Gateway (DNS or
"ip host"
commands).
Note
|
If ASR and
TTS use the same server, the MRCP server might allocate one license for the ASR
session and a second license for the TTS section. If you are hosting both ASR
and TTS on the same speech server, you must select the
ASR/TTS
use the same MRCP server
option in the IVR Service configuration tab in the
Operations Console and follow the instructions in the step below.
|
Do one of the
following:  
-
If you are using ACE, the server name is configured to the virtual IP (VIP) of the Call Server on ACE. For more information,
see the Configure High Availability for Unified CVP section.
-
The
primary and backup servers must be configured. If using name resolution local
to the Gateway (rather than DNS) specify:
ip host
asr-
<locale> <ASR server for
locale>
ip host
asr-
<locale>-backup <backup ASR server
for locale>
ip host
tts-
<locale> <TTS server for
locale>
ip host
tts-
<locale>-backup <backup TTS server
for locale>
Example
for English US, use:
ip host
asr-en-us 10.86.129.215
|
Step 12 |
If you want
the ASR and TTS to use the same MRCP server option, you must configure the
gateway as follows.
-
In the
IVR Service in the Operations Console, select the
ASR/TTS
use the same MRCP server option.
-
Add the
following two host names to the gateway configuration:
-
ip
host asrtts-
<locale>
<IP Address Of MRCP Server>
-
ip
host asrtts-
<locale>
-backup <IP Address Of MRCP
Server>
Where the
locale might be something like en-us or es-es,
resulting in
asrtts-en-us or
asrtts-es-es .
-
Change
the 'ivr asr-server' and 'ivr tts-server' lines as follows for MRCPV1:
-
Change
the 'ivr asr-server' and 'ivr tts-server' lines as follows for MRCPV2:
|
Step 13 |
Configure
the speech servers to work with Unified CVP.
Caution
|
The
Operations Console can only manage speech servers installed on
Windows, not on Linux. If the speech server is installed on
Linux, the server cannot be managed.
|
To ensure
that the speech servers work with Unified CVP, you must make the following
changes on each speech server as part of configuring the Unified CVP solution.
If you are
using Nuance SpeechWorks MediaServer (SWMS), the configuration file is
osserver.cfg. If you are using Nuance Speech Server (NSS), the configuration
file is NSSserver.cfg.
Make the
following changes to the Nuance configuration file:
-
Change:
server.resource.2.url VXIString media/speechrecognizer
To:
server.resource.2.url VXIString recognizer
-
Change:
server.resource.4.url VXIString media/speechsynthesizer
To:
server.resource.4.url VXIString synthesizer
-
Change:
server.mrcp1.resource.3.url VXIString media/speechrecognizer
To:
server.mrcp1.resource.3.url VXIString /recognizer
-
Change:
server.mrcp1.resource.2.url VXIString media/speechsynthesizer
To:
server.mrcp1.resource.2.url VXIString media/synthesizer
-
Change:
server.mrcp1.transport.port VXIInteger 4900
To:
server.mrcp1.transport.port VXIInteger 554
If you are using Nuance Speech Server 5 and Nuance Vocalizer for Network 5, then make changes to the configuration files
for each application. Make the following changes to the Nuance Speech Server 5 configuration file (NSSserver.cfg):
-
Change:
server.mrcp1.resource.3.url VXIString media/speechrecognizer
To:
server.mrcp1.resource.3.url VXIString /recognizer
-
Change:
server.mrcp1.resource.2.url VXIString media/speechsynthesizer
To:
server.mrcp1.resource.2.url VXIString /synthesizer
-
Change:
server.mrcp1.transport.port VXIInteger 4900
To:
server.mrcp1.transport.port VXIInteger 554
-
Change:
server.mrcp1.transport.dtmfPayloadType VXIInteger 96
To:
server.mrcp1.transport.dtmfPayloadType VXIInteger 101
-
Uncomment the
following: server.rtp.dtmfTriggerLeading VXIInteger 0
If you
are using the Nuance Vocalizer for Network 5 TTS System, the following
configuration files will need to be updated:
<install path>\Nuance Vocalizer for Network 5.0\config\ttsrshclient.xml
-
Change:
<ssml_validation>strict</ssml_validation>
To:<ssml_validation>warn</ssml_validation>
<install path>\Nuance Vocalizer for Network 5.0\config\ttssapi.xml
-
Change:
<ssml_validation>strict</ssml_validation>
To:
<ssml_validation>warn</ssml_validation>
If you are using Nuance Recognizer 10.0 and Nuance Speech Server 6.2, make the following changes to the Nuance configuration
file (NSSserver.cfg - C:\Program Files (x86)\Nuance\Speech Server\Server\config):
-
Change: server.mrcp1.resource.3.url VXIString
media/speechrecognizer
To:
server.mrcp1.resource.3.url VXIString /recognizer
-
Change:
server.mrcp1.resource.2.url VXIString media/speechsynthesizer
To:
server.mrcp1.resource.2.url VXIString /synthesizer
-
Change: server.mrcp1.transport.port VXIInteger 4900
To:
server.mrcp1.transport.port VXIInteger 554
-
Change:
server.mrcp1.transport.dtmfPayloadType VXIInteger 96
To: server.mrcp1.transport.dtmfPayloadType VXIInteger
Make the
following change to the Baseline.xml file
C:\Program Files\Nuance\Recognizer\config
Change:
<ssml_validation>strict</ssml_validation>
To:<ssml_validation>warn</ssml_validation>.
If you are using Nuance Recognizer 10.5 and Nuance Speech Server 6.5, then refer to the relevant Nuance Speech Suite Install
Guide available at https://network.nuance.com/portal/server.pt/directory/nuance_speech_suite_10_5/16535.
|
Step 14 |
Configure
SIP-Specific Actions.
On the
Unified CM
server, CCMAdmin Publisher, configure
SIP-specific actions:
-
Create
SIP trunks:
-
If
you are using a SIP Proxy Server, set up a SIP trunk to the SIP Proxy Server.
-
Add
a SIP Trunk for the Unified CVP Call Server.
-
Add
a SIP Trunk for each Ingress gateway that will send SIP calls to Unified CVP
that might be routed to
Unified CM.
Select
and add the following:
-
Trunk Type:
SIP trunk
-
Device Protocol:
SIP
-
Destination Address: IP address or host name of the SIP Proxy Server (if using
a SIP Proxy Server). If not using a SIP Proxy Server, enter the IP address or
host name of the Unified CVP Call Server.
-
DTMF Signaling Method:
RFC 2833
-
Do
not check the
Media Termination Point Required checkbox.
-
If
you are using UDP as the outgoing transport on Unified CVP, also set the
outgoing transport to
UDP on the SIP Trunk Security Profile.
-
Add
route patterns for outbound calls from
Unified CM
devices using a SIP Trunk to the Unified CVP Call Server. Also, add a route
pattern for error DN.
Note
|
CVP
solution does not support 100rel. On the SIP profile for the Trunk, confirm
that SIP Rel1xx Options are disabled.
For
warm transfers, the call from Agent 1 to Agent 2 does not typically use a SIP
Trunk, but you must configure the CTI Route Point for that dialed number on the
Unified CM
Server and associate that number with your peripheral gateway user (PGUSER) for
the JTAPI gateway on the
Unified CM
peripheral gateway. An alternative is to use the Dialed Number Plan on
Unified ICME to
bypass the CTI Route Point.
|
-
Select
.
-
Route Pattern: Specify the route pattern; for example: 3xxx for a TDM phone
that dials 9+3xxx and all
Unified ICME
scripts are set up for 3xxx dialed numbers.
-
Gateway/Route List: Select the SIP Trunk defined in Step 2.
-
If you
are sending calls to
Unified CM
using an SRV cluster domain name, configure the cluster domain name.
|
Step 15 |
(Optional) Configure
the
SIP Proxy
Server.
From the
CUSP Server Administration web page (http://<CUSP server>/admin):
-
Configure the SIP static routes to the Unified CVP Call Server(s),
Unified CM SIP
trunks, and Gateways.
Configure the SIP static routes for intermediary transfers for ring tone, playback dialed numbers, and error playback dialed
numbers.
Note
|
For failover and load balancing of calls to multiple destinations, configure the CUSP Server static route with priority and
weight.
|
See the SIP Devices Configuration and SIP Dialed Number Pattern Matching Algorithm for detailed information.
-
Configure Access Control Lists for Unified CVP calls.
-
Select
.
-
Set
address pattern:
all
-
Configure the service parameters.
Select
Service Parameters, and set the following:
-
Add
record route:
off
-
Maximum invite retransmission count:
2
-
Proxy Domain and Cluster Name: if using DNS SRV, set to the FQDN of your Proxy
Server SRV name.
-
Write
down the IP address and host name of the SIP Proxy Server. You need this
information when configuring the SIP Proxy Server in Unified CVP.
-
If
using redundant SIP Proxy Servers (primary and secondary or load balancing),
decide whether to use DNS server lookups for SRV records or non-DNS based local
SRV record configuration.
The
Comprehensive call flow model with SIP calls will typically be deployed with
dual CUSP Servers for redundancy. In some cases, you might want to purchase a
second CUSP Server. Regardless, the default transport for deployment will be
UDP. Make sure you
always set the AddRecordRoute setting to
Off with CUSP Servers.
Configure the SRV records on the DNS server or locally on Unified CVP with an
.xml file (local xml configuration avoids the overhead of DNS lookups with each
call).
|
Step 16 |
Configure
Peripheral Gateways (PGs).
On the NAM,
ICM Configuration Manager,
PG
Explorer tool, configure a peripheral gateway (PG) for the Unified CVP.
Configure a PG for each Unified CVP Call Server as follows:
In the tree
view pane, select the applicable PG.
Logical
Controller tab:
Peripheral
tab:
-
Peripheral Name: Descriptive name of this Unified CVP peripheral
For
example:
<location>_<cvp1> or <dns_name>
-
Client
Type:
VRU
-
Select:
Enable Post-routing
Advanced tab:
Routing
Client tab:
-
Name: By
convention, use the same name as the peripheral
-
Client
Type:
VRU
-
If you
are in a
Unified ICMH
environment and configuring the CICM, then do the following:
Note
|
If you are using a VXML gateway that is not co-located, then configure the following dial-peer to handle the error case:
Example:
dial-peer voice 9292 voip
description SIP error dial-peer
session protocol sipv2
session target ipv4:<destination IP_address for the VXML gateway>
session transport tcp
codec g711ulaw
destination-pattern 929292T
dtmf-relay rtp-nte
no vad
This may vary depending on the type of deployment.
|
|