Gateway Configuration

Configure Gateway

Procedure


Step 1

Log in to Operations Console and click Device Management > Gateway.

The Find, Add, Delete, Edit Gateways window opens.

Step 2

Click Add New.

Note

 

To use an existing Gateway as a template for configuring a new Gateway, select a Gateway from the list of available Gateways and click Use As Template and perform Steps 3 to 5.

Step 3

Click the General tab, enter the field values, and click Save. See General Settings.

Step 4

(Optional) Click the Device Pool tab, enter the field values, and click Save. See Add or Remove Device From Device Pool.

Step 5

Click Save.

Step 6

(Optional) If the call control client placed the Correlation ID in a GTD parameter other than uus.dat, specify the following parameters to configure a gateway to enable incoming UUI to be used as the Correlation ID.


                            conf t
                            application
                            service <your-cvp-service-name>
                            param use-uui-as-corrid Y (Refer to Note 1)
                            param correlation-gtd-attribute XXX (Refer to Note 2)
                            param correlation-gtd-instance N (Refer to Note 2)
                            param correlation-gtd-field YYY (Refer to Note 2)
                            dial-peer voice 123 pots
                            service <your-cvp-service-name>
                        

Gateway Settings

General Settings

After adding an IOS Gateway, you can run a subset of IOS Gateway commands on the Gateway from the Operations Console.

The Ingress Gateway is the point at which an incoming call enters the Unified CVP solution. It terminates Time Division Multiplexing (TDM) phone lines on one side and implements VoIP on the other side. It also provides for sophisticated call routing capabilities at the command of other Unified solution components. It works with SIP and also supports Media Gateway Control Protocol (MGCP) for use with Unified CM.

The service configuration parameters for the Call Server host and port are meant for the VRU-Only call flow model for IOS VoiceXML Gateway. These parameters are optional and you can use them to override the IP address or port number of the Call Server that comes through the SIP app-info header.

An Egress Gateway is typically used in Call Director model to provide access to a call center automatic call distributor (ACD) or third-party IVR.

To configure General settings on a Gateway, on the General tab, enter the field values, as listed in the following table:

Table 1. Unified ICM—General Tab Configuration Settings

Field

Description

Default

Value

Restart Required

IP Address

The IP address of a Unified ICM Server

None

Valid IP address

No

Hostname

The name of the Unified ICM Server

None

Valid DNS name. It includes alphanumeric characters and a dash.

No

Description

Additional information of the Unified ICM Server

None

Up to 1024 characters

No

Device Admin URL

The URL for the Unified ICM Web configuration application.

None

Valid URL

No

Activate Gateway Configuration

Activate the gateway configuration by entering these commands:

Procedure

Step 1

call application voice load CVPSelfService

Step 2

call application voice load HelloWorld


Configure Gateway Settings for Comprehensive Call Flow Model

Procedure


Step 1

Install the IOS image on the Ingress Gateway.

For detailed information, see the Cisco IOS documentation.

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:

  1. 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
    
  2. Configure the DNS zone file for the separate DNS server that displays how the Service (SRV) records are configured.

    Note

     

    SRV with DNS can be used in any of the SIP call flow models, with or without a Proxy server. Standard A type DNS queries can be used as well for the calls, without SRV, but they lose the load balancing and failover capabilities.

    See DNS Zone File Configuration for Call Director Call Flow Model.

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.

See the SIP Devices Configuration and the Operations Console Online Help, Managing devices > Configuring a SIP Proxy Server for details.

Step 8

Configure SIP-Specific Actions.

On the Unified CM server, CCMAdmin Publisher, configure SIP-specific actions:

  1. 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 Device > Trunk > Add New 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.

  2. 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.

  3. Select Call Routing > Route/Hunt > Route Pattern > Add New.

    • 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.

  4. If you are sending calls to Unified CM using an SRV cluster domain name, configure the cluster domain name.

    • Select: Enterprise Parameters > Clusterwide Domain Configuration.

    • Add the Cluster fully qualified domain name: FQDN.

For detailed instructions about using Unified CM and the CUSP Server, see the CUSP Server documentation.

Step 9

(Optional) Configure the SIP Proxy Server.

From the CUSP Server Administration web page (http://<CUSP server>/admin):

  1. 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.

  2. Configure Access Control Lists for Unified CVP calls.

    • Select Proxy Settings > Incoming ACL.

    • Set address pattern: all

  3. 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.

  4. 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.

  5. 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 10

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:

  • Client Type: VRU

  • Name: A name descriptive of this PG

    For example: <location>_A for side A of a particular location

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:

  • Select the name of the Unified CVP VRU from the Network VRU field drop-down list.

    For example: cvpVRU

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:

    • Do not select the Network Transfer Preferred checkbox

    • Routing client: INCRP NIC


Configure Gateway Settings for Call Director Call Flow Model

Procedure


Step 1

Perform Steps 1 to 4 of the Configure Gateway Settings for Comprehensive Call Flow Model procedure.

Step 2

Configure the Ingress Gateway:

  1. Configure the Ingress Gateway dial-peer for the Unified CVP Call Server.

  2. Configure a dial-peer for ringtone and error.

  3. If you are using a Proxy Server, configure your session target in the outbound dial peer to point to the Proxy Server.

  4. If you are using the sip-server global configuration, then configure the sip-server in the sip-ua section to be your Proxy Server and point the session target of the dial-peer to the sip-server global variable.

Note

 

Make sure your dial plan includes this information. You will need to see the Dial plan when you configure the SIP Proxy Server for Unified CVP.

The SIP Service voip dial peer and the destination pattern on the Ingress Gateway must match the DNIS in static routes on the SIP Proxy Server or Unified CVP Call Server.

Step 3

For SIP without a Proxy Server, complete the following steps:

  1. If you are using DNS query with SRV or A types from the gateway, configure the gateway to use DNS.

    See the SIP Devices Configuration and Operations Console online help for detailed instructions. If you are using DNS query with SRV or A types from the gateway, use the gateway configuration CLI as shown below:

    Non-DNS Setup:

    
    sip-ua
    sip-server ipv4:xx.xx.xxx.xxx:5060
    !
    

    DNS Setup:

    
    ip domain name patz.cisco.com
    ip name-server 10.10.111.16
    !
    sip-ua
    sip-server dns:cvp.pats.cisco.com
    !
    
  2. Configure the DNS zone file for the separate DNS server that displays how the Service (SRV) records are configured.

    Note

     

    SRV with DNS can be used in any of the SIP call flow models, with or without a Proxy server. Standard A type DNS queries can be used as well for the calls, without SRV, but they lose the load balancing and failover capabilities.

    See the DNS Zone File Configuration for Call Director Call Flow Model for more information.

Step 4

For SIP with a Proxy Server, use one of the following methods:

Note

 

You can 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.

If you are using the DNS Server, you can set your SIP Service as the Host Name (either A or SRV type).

Step 5

On the Unified CM server, CCMAdmin Publisher, complete the following SIP-specific actions:

  1. 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.

    To add an SIP trunk, select Device > Trunk > Add New and use the following parameters:

    • 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 check box.

    • 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.

    • Connection to CUSP Server: use 5060 as the default port.

  2. Add route patterns for outbound calls from the Unified CM devices using a SIP Trunk to the Unified CVP Call Server. Also, add a route pattern for error DN.

    Select Call Routing > Route/Hunt > Route Pattern > Add New

    Add the following:

    • 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 the previous substep.

    Note

     

    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.

  3. If you are sending calls to Unified CM using an SRV cluster domain name, select Enterprise Parameters > Clusterwide Domain Configuration and add the Cluster fully qualified domain name FQDN.

Step 6

(Optionally) Configure the SIP Proxy Server.

  1. Configure the SIP static routes to the Unified CVP Call Servers, Unified CM SIP trunks, and Gateways.

    Configure the SIP static routes for intermediary transfers for ringtone, 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.

  2. Configure Access Control Lists for Unified CVP calls.

    Select Proxy Settings > Incoming ACL.

    Address pattern: all

  3. Configure the service parameters.

    Select Service Parameters, then 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

  4. 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.)

  5. If using redundant SIP Proxy Servers (primary and secondary or load balancing), then decide whether to use DNS server lookups for SRV records or non-DNS based local SRV record configuration.

    Note

     

    If a single CUSP Server is used, then SRV record usage is not required.

    Configure the SRV records on the DNS server or locally on Unified CVP with a .xml file (local xml configuration avoids the overhead of DNS lookups with each call).

    Note

     

    See the Local SRV File Configuration Example for SIP Messaging Redundancy section for details.

    The Call Director 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 disable the record-route in a CUSP server as this advanced feature is not supported in Contact Center deployments.

    For the required settings in the Unified CM Publisher configuration, see Cisco Unified SIP Proxy documentation.

Step 7

Configure the PGs for the switch leg.

On Unified ICME, ICM Configuration Manager, PG Explorer tool:

  1. Configure each peripheral gateway (PG) to be used for the Switch leg. In the tree view pane, select the applicable PG, and set the following:

    1. Logical Controller tab:

      • Client Type: VRU

      • Name: A name descriptive of this PG

        For example: <location>_A for side A of a particular location

    2. Peripheral tab:

      • Peripheral Name: A name descriptive of this Unified CVP peripheral

        For example: <location>_<cvp1> or <dns_name>

      • Client Type: VRU

      • Select the check box: Enable Post-routing

    3. Routing Client tab:

      • Name: By convention, use the same name as the peripheral.

      • Client Type: VRU

    For more information, see the ICM Configuration Guide for Cisco ICM Enterprise Edition.

  2. Configure a peripheral for each Unified CVP Call Server to be used for a Switch leg connected to each peripheral gateway.


Configure Gateway Settings for VRU-Only Call Flow Model: Type 8

Procedure


Step 1

Using the Unified CVP Operations Console or the Unified CVP product CD, transfer the following script, configuration, and .wav files to the VoiceXML Gateway used for the VRU leg. Perform Step 2 of the Configure Gateway Settings for Comprehensive Call Flow Model procedure.

Step 2

Configure the VXML gateway base settings.

Step 3

Configure the VXML gateway service settings.

Step 4

Configure the ICM service.

Using the Operations Console, select Device Management > CVP Call Server > ICM tab. On each Unified CVP Call Server, configure the ICM Service by specifying the following required information:

  1. VRU connection port number.

    Set the VRU Connection Port to match the VRU connection Port defined in ICM Setup for the corresponding VRU peripheral gateway (PIM).

  2. Maximum Length of DNIS.

    Set the maximum length DNIS to a number which is at least the length of the translation route DNIS numbers.

    Example: if the Gateway dial pattern is 1800******, the maximum DNIS length is 10.

  3. Call service IDs: New Call and Pre-routed.

    Enter the new and pre-routed call service IDs. Configure the ports for both groups according to the licenses purchased, call profiles, and capacity by completing the required fields on this tab.

  4. Trunk group IDs: New Call and Pre-routed.

    • Enter the new and pre-routed call trunk group IDs

    • Configure the group number for the Pre-routed Call Trunk group. The group number must match the trunk group number in the Network Trunk group used for the translation route

    • Configure the number of ports according to the licenses purchased and capacity

    • Configure each of the numbers used for translation routes. (The "New Call" group is not used since the calls are being sent to the VRU (Unified CVP) after some initial processing by the NIC/Unified ICME)

  5. Dialed numbers used in the translation route.

    Add the dialed numbers in the DNIS field.

  6. Check the default values of the other settings and change, if desired.


VoiceXML Gateway Configuration Examples

Example Gateway Settings for Type 8 Call Flow Model
The first part of the following example provides the basic configuration for setting a VoiceXML gateway:
  • Applies a timestamp to debugging and log messages

  • Turns on logging

  • Turns off printing to the command line interface console

  • Sends RTP packets

  • Configures ASR/TTS Server

  • Configures gateway settings

The last part of this example provides the following:
  • Initiates the VoiceXML leg

  • Plays a .wav file that enables caller to hear message from critical_error.wav

  • Logs errors on the gateway when the call fails

service timestamps debug datetime msec
service timestamps log datetime msec
service internal
logging buffered 99999999 debugging
no logging console
ip cef
no ip domain lookup
ip host tts-en-us <IP of TTS or MRCP Server>
ip host asr-en-us <IP of ASR or MRCP Server>
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
signaling forward unconditional
h323
sip
min-se 360
header-passing
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
ivr prompt memory 15000
ivr prompt streamed none
ivr asr-server rtsp://asr-en-us/recognizer
ivr tts-server rtsp://tts-en-us/synthesizer
mrcp client timeout connect 10
mrcp client timeout message 10
mrcp client rtpsetup enable
rtsp client timeout connect 10
rtsp client timeout message 10
vxml tree memory 500
http client cache memory file 500
http client connection timeout 60
http client response timeout 30
http client connection idle timeout 10
gateway
timer receive-rtcp 6
!
ip rtcp report interval 3000
application
service new-call flash:bootstrap.vxml
service cvperror flash:cvperror.tcl
service handoff flash:handoff.tcl
Example of Dial-peer for ICM VRU Label for Type 8 Call Flow Model
The following example provides the configuration for an ICM VRU label dial-peer for the Type8 Unified CVP VRU-Only call flow model:

dial-peer voice 777 voip
 description ICM VRU label
 service bootstrap
 voice-class codec 1
 incoming called-number <your sendtovru label pattern here> 
 dtmf-relay rtp-nte
 no vad
 !

Configure Gateway Settings for VRU-Only: Type 7

Procedure


Step 1

Transfer the following script, configuration, and .wav files to the VoiceXML Gateway used for the VRU leg, using the Unified CVP Operations Console. Perform Step 2 of the Configure Gateway Settings for Comprehensive Call Flow Model procedure.

Step 2

Configure the VoiceXML gateway base settings.

Step 3

Configure the VoiceXML gateway service settings.

Step 4

Configure the ICM Service for each Call Server.

In the Operations Console, select Device Management > CVP Call Server > ICM tab. For each Unified CVP Call Server, configure the ICM Service by specifying the following required information:

  1. VRU connection port number.

    Set the VRU Connection Port to match the VRU connection Port defined in ICM Setup for the corresponding VRU peripheral gateway (PIM).

  2. Set the maximum length DNIS to the length of the Network Routing Number.

    Example: if the Gateway dial pattern is 1800******, the maximum DNIS length is 10.

  3. Call service IDs: New Call and Pre-routed.

    Enter the new and pre-routed call service IDs. Configure the ports for both groups according to the licenses purchased, call profiles, and capacity by completing the required fields on this tab

  4. Trunk group IDs: New Call and Pre-routed.

    Enter the new and pre-routed call trunk group IDs. Configure the group number for the Pre-routed Call Trunk group. The group number must match the trunk group number in the Network Trunk group used for the translation route.

    Configure the number of ports according to the licenses purchased and capacity. Configure each of the numbers used for translation routes. (The "New Call" group is not used since the calls are being sent to the VRU (Unified CVP) after some initial processing by the NIC/Unified ICME.)

  5. Check the default values of other settings and change, if desired.


VoiceXML Gateway Configuration: Example Gateway Settings for Type 7


Note


While using ASR/TTS, use a single version of MRCP (v1/v2) instead of using it in mixed mode.


The first part of the following example provides the basic configuration for setting a VoiceXML gateway:

  • Applies a timestamp to debugging and log messages

  • Turns on logging

  • Turns off printing to the command line interface console

  • Sends RTP packets

  • Configures ASR/TTS Server

  • Configures gateway settings

The last part of this example provides the following:

  • Initiates the VoiceXML leg

  • Plays a .wav file that enables caller to hear message from critical_error.wav

  • Logs errors on the gateway when the call fails

    service timestamps debug datetime msec
    service timestamps log datetime msec
    service internal
    logging buffered 99999999 debugging
    no logging console
    ip cef
    no ip domain lookup
    ip host tts-en-us <IP of TTS or MRCP Server>
    ip host asr-en-us <IP of ASR or MRCP Server>
    voice rtp send-recv
    !
    voice service voip
     allow-connections h323 to h323
     signaling forward unconditional
     h323
     sip
     min-se 360
     header-passing
    voice class codec 1
     codec preference 1 g711ulaw
     codec preference 2 g729r8
    !
    ivr prompt memory 15000
    ivr prompt streamed none
    ivr asr-server rtsp://asr-en-us/recognizer
    ivr tts-server rtsp://tts-en-us/synthesizer
    mrcp client timeout connect 10
    mrcp client timeout message 10
    mrcp client rtpsetup enable
    rtsp client timeout connect 10
    rtsp client timeout message 10
    vxml tree memory 500
    http client cache memory pool 15000
    http client cache memory file 500
    http client connection timeout 60
    http client response timeout 30
    http client connection idle timeout 10
    gateway
     timer receive-rtcp 6
    !
    ip rtcp report interval 3000
    application
     service new-call flash:bootstrap.vxml
     service cvperror flash:cvperror.tcl
     service handoff flash:handoff.tcl
     service bootstrap flash:bootstrap.tcl
     !

The following example provides the configuration for an ICM VRU label dial-peer for the Type 7 Unified CVP VRU-Only call flow model:

dial-peer voice 777 voip
 description ICM VRU label
 service bootstrap
 voice-class codec 1
 incoming called-number <your sendtovru label pattern here>
 dtmf-relay rtp-nte
 no vad
 !

Transfer Script and Media File to Gateway

Transfer a single script or media file at a time from the Operations Console.

Procedure


Step 1

Log in to the Operations Console and from the Device Management menu, select the type of server to which to transfer the script file.

Example:

To transfer a script or a media file to a Gateway, select Device Management > Gateway..

The Find, Add, Delete, Edit window lists any servers that have been added to the Operations Console.

Step 2

Select a server by clicking the link in its Hostname field or by clicking the radio button preceding it and then clicking Edit.

Step 3

Select File Transfer in the toolbar, and then click Scripts and Media.

The Scripts and Media File Transfer page appears, listing the host name and IP address for the selected device. Script and Media files currently stored in the Operations Server database are listed in the Select From available Script Files drop box.

Step 4

If the script or media file is not listed in the Select From Available Script Files drop box:

  1. Click Select a Script or Media File from Your Local PC.

  2. Enter the file name in the text box or click Browse to search for the script or media file on the local file system.

Step 5

If the script or media file is listed in the Select From Available Script Files drop box, select the script or media file.

Step 6

Click Transfer to send the file to the device.


Configure Gateway Settings to modify Outgoing SIP Header

In some scenarios a gateway may need to interact with third-party ACDs, which requires modifications to certain SIP headers.

In this example, the SIP Refer-To header in the outgoing SIP REFER message is modified to pass the Application-to-Application information (AAI) as User-to-User information (UUI):

  1. Create SIP-Copylist.

    voice class sip-copylist 1
     sip-header Refer-To
    
  2. Create SIP-Profile.

    voice class sip-profiles 1
     request REFER peer-header sip Refer-To copy ";(.*)" u01 
     request REFER sip-header Refer-To modify "Refer-To: (.*)" "Refer-To: <\1;\u01” 
    
  3. Profile at the dial peer level:
    1. Copy-list to be applied under the inbound dial-peer: voice-class sip copy-list 1.
    2. SIP-profiles to be applied under the outbound dial-peer: voice-class sip profiles 1.
    
    Example:
    ==================================================================================================
    1. Example of SIP copy-list:
    voice class sip-copylist 2
    sip-header Refer-To
     
    2. Example of SIP Profile:
    voice class sip-profiles 2
    request REFER peer-header sip Refer-To copy ";aai=(.*)" u02
    request REFER sip-header Refer-To modify "Refer-To: (.*)" "Refer-To: <\1;User-to-User=\u02"
     
    3. Example of dial peer:
     
    Dial peer to fwd the call to VVB
    -----------------------------------------
    dial-peer voice 345679 voip
    destination-pattern 369852147
    session protocol sipv2
    session target ipv4:10.78.0.94
    session transport tcp
    voice-class sip copy-list 2
    codec g711ulaw
     
    Dial peer to modify the outgoing SIP REFER message
    --------------------------------------------------------------------
    dial-peer voice 345678 voip
    session protocol sipv2
    session target sip-server
    session transport tcp
    incoming called-number 369852147
    voice-class sip profiles 2
    codec g711ulaw
    

Once the aforesaid configuration is done, the SIP REFER message with the following SIP header:

Refer-To: <sip:1247@72.163.166.103;aai=PD%2C04%3BC8%2C48656c6c6520746865726521%3BFA%2C00001016421311070956>

is modified to:

Refer-To: <sip:1247@72.163.166.103;User-to-User=PD%2C04%3BC8%2C48656c6c6520746865726521%3BFA%2C00001016421311070956>

With similar configuration on gateway, other SIP headers can also be modified according to the call flow needs.