Configure Cisco IOS Enterprise Voice Gateway

About Ingress and VXML Gateway Configuration

Complete the following procedures to configure the Ingress Gateway and VXML Gateway. Instructions are applicable to both TDM and Cisco Unified Border Element (CUBE) Voice gateways, unless otherwise noted.

You may also have Cisco Virtualized Voice Browser (Cisco VVB) as part of your deployment. For information about Cisco VVB, see Install and Configure Cisco Virtualized Voice Browser.


Note

Complete all configuration steps in enable > configuration terminal mode.

Common Configuration for the Ingress Gateway and VXML Gateway 

logging buffered 2000000 debugging
no logging console	
service timestamps debug datetime msec localtime
ip routing
ip cef	
ip source-route	
interface GigabitEthernet0/0
     ip route-cache same-interface
     duplex auto
     speed auto
     no keepalive
     no cdp enable

voice service voip
          ip address trusted list
          ipv4 0.0.0.0 0.0.0.0 # OR an explicit Source IP Address Trust List
     allow-connections sip to sip
    	signaling forward unconditional

Configure Ingress Gateway

Procedure

Step 1

Configure global settings. 

voice service voip
	 allow-connections sip to sip
	signaling forward unconditional
	# If this gateway is being licensed as a Cisco UBE the following lines are also required
 mode border-element
 ip address trusted list
  ipv4 0.0.0.0 0.0.0.0               # Or an explicit Source IP Address Trust List
sip
		rel1xx disable
		header-passing
		options-ping 60
  midcall-signaling passthru
Step 2

Configure voice codec preference: 

voice class codec 1
     codec preference 1 g711ulaw
     codec preference 2 g711alaw
     codec preference 3 g729r8
Step 3

Configure default services: 

#Default Services
application
     service survivability flash:survivability.tcl
Step 4

Configure gateway and sip-ua timers: 

gateway
	media-inactivity-criteria all
	timer receive-rtp 1200

sip-ua
	retry invite 2
	retry bye 1
	timers expires 60000
	timers connect 1000
	reason-header override
Step 5

Configure POTS dial-peers: 

# Configure Unified CVP survivability
dial-peer voice 1 pots
     description CVP TDM dial-peer
     service survivability
     incoming called-number .T
     direct-inward-dial
Note 

This is required for TDM gateways only.

Step 6

Configure the switch leg: 

#Configure the Switch leg where
# preference is used to distinguish between sides.
# max-conn is used prevent overloading of Unifed CVP
# options-keepalive is used to handle failover
# Note: the example below is for gateways located on the A-side of a geographically
#distributed deployment
# Note: Ensure that you configure switch dial-peers for each Unified CVP server.

dial-peer voice 70021 voip
     description Used for Switch leg SIP Direct
     preference 1
     max-conn 225
     destination-pattern xxxx...... #Customer specific destination pattern
     session protocol sipv2
     session target ipv4:###.###.###.###     #IP Address for Unified CVP, SideA
     session transport tcp
     voice-class codec 1
     voice-class sip options-keepalive up-interval 12 down-interval 65 retry 2
     dtmf-relay rtp-nte
     no vad

dial-peer voice 70022 voip
     description Used for Switch leg SIP Direct
     preference 2
     max-conn 225
     destination-pattern xxxx...... #Customer specific destination pattern
     session protocol sipv2
     session target ipv4:###.###.###.###    #IP Address for Unified CVP, SideB
     session transport tcp
     voice-class codec 1
     voice-class sip options-keepalive up-interval 12 down-interval 65 retry 2
     dtmf-relay rtp-nte
     no vad
Step 7

Configure the hardware resources (transcoder, conference bridge, and MTP):

Note 

This configuration section is unnecessary for virtual CUBE or CSR 1000v Gateways. They do not have physical DSP resources. 

#For gateways with physical DSP resources, configure Hardware resources using 
#Unified Communications Domain Manager.

# Configure the voice-cards share the DSP resources located in Slot0
voice-card 0
     dspfarm
     dsp services dspfarm
voice-card 1
     dspfarm
     dsp services dspfarm
voice-card 2
     dspfarm
     dsp services dspfarm
voice-card 3
     dspfarm
     dsp services dspfarm
voice-card 4
     dspfarm	
     dsp services dspfarm

# Point to the contact center call manager
sccp local GigabitEthernet0/0
     sccp ccm ###.###.###.### identifier 1 priority 1 version 7.0 # Cisco Unified CM sub 1
     sccp ccm ###.###.###.### identifier 2 priority 2 version 7.0 # Cisco Unifed CM sub 2

# Add a SCCP group for each of the hardware resource types
sccp ccm group 1
     associate ccm 1 priority 1
     associate profile 2 register <gatewaynamemtp>
     associate profile 1 register <gatewaynameconf>
     associate profile 3 register <gatewaynamexcode>

# Configure DSPFarms for Conference, MTP and Transcoder

dspfarm profile 1 conference
     codec g711ulaw
     codec g711alaw
     codec g729r8
     maximum sessions 24
     associate application SCCP

dspfarm profile 2 mtp
     codec g711ulaw
     codec g711alaw
     codec g729r8
     maximum sessions software 500
     associate application SCCP

dspfarm profile 3 transcode universal
     codec g711ulaw
     codec g711alaw
     codec g729r8
     maximum sessions 52
     associate application SCCP
Step 8

Optional, configure the SIP Trunking: 

# Configure the resources to be monitored
voice class resource-group 1
     resource cpu 1-min-avg threshold high 80 low 60
     resource ds0
     resource dsp
     resource mem total-mem
     periodic-report interval 30

# Configure one rai target for each CVP Server
sip-ua
     rai target ipv4:###.###.###.### resource-group1 # CVPA
     rai target ipv4:###.###.###.### resource-group1 # CVPB
     permit hostname dns:%Requires manual replacement - ServerGroup Name defined in CVP.System.SIP Server Groups%
Step 9

Configure incoming PSTN SIP trunk dial peer: 

dial-peer voice 70000 voip
     description Incoming Call From PSTN SIP Trunk
     service survivability 
     incoming called-number xxxx……    # Customer specific incoming called-number pattern
     voice-class sip rel1xx disable
     dtmf-relay rtp-nte 
     session protocol sipv2
     voice-class codec 1
     no vad
Note 

This is required for CUBE only.

Configure VXML Gateway

Before you begin


Note

If you have configured VVB, it is not mandatory to configure VXML Gateway. You may configure either VVB or VXML Gateway, or configure both.

Procedure

Step 1

Configure global settings: 

voice service voip
	 allow-connections sip to sip
	signaling forward unconditional
	# If this gateway is being licensed as a Cisco UBE the following lines are also required
 mode border-element
 ip address trusted list
  ipv4 0.0.0.0 0.0.0.0               # Or an explicit Source IP Address Trust List
sip
		rel1xx disable
		header-passing
		options-ping 60
  midcall-signaling passthru
Step 2

Configure default Unified CVP services: 

#Default Unified CVP Services
application
     service new-call flash:bootstrap.vxml
     service CVPSelfService flash:CVPSelfServiceBootstrap.vxml
     service ringtone flash:ringtone.tcl
     service cvperror flash:cvperror.tcl
     service bootstrap flash:bootstrap.tcl
     service handoff flash:handoff.tcl
Step 3

Configure dial-peers:

Note 

While configuring VXML gateway voice class codec must not be used. G711ulaw may be used in general for the dial-peers, but still depending on the implementation the other codec may be used. 

# Configure Unified CVP Ringtone
dial-peer voice 919191 voip
     description CVP SIP ringtone dial-peer
     service ringtone
     incoming called-number 9191T
     voice-class sip rel1xx disable
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad

# Configure Unified CVP Error
dial-peer voice 929292 voip
     description CVP SIP error dial-peer
     service cvperror
     incoming called-number 9292T
     voice-class sip rel1xx disable
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
Step 4

Configure default Unified CVP HTTP, ivr, rtsp, mrcp and vxml settings: 

http client cache memory pool 15000
http client cache memory file 1000
http client cache refresh 864000
no http client connection persistent
http client connection timeout 60
http client connection idle timeout 10
http client response timeout 30
ivr prompt memory 15000

vxml tree memory 500
vxml audioerror
vxml version 2.0
Step 5

Configure VXML leg where the incoming called-number matches the Network VRU Label: 

dial-peer voice 7777 voip
     description Used for VRU leg
     service bootstrap
     incoming called-number 777T
     dtmf-relay rtp-nte
     codec g711ulaw
     no vad
Step 6

Exit configuration mode and use the Cisco IOS CLI command call application voice load <service_Name> to load the transferred Unified CVP files into the Cisco IOS memory for each Unified CVP service:

  • call application voice load new-call

  • call application voice load CVPSelfService

  • call application voice load ringtone

  • call application voice load cvperror

  • call application voice load bootstrap

  • call application voice load handoff

Configure Codec for Ingress and VXML Gateways

Configure Ingress Gateway

Procedure

Step 1

Add the voice class codec 1 to set the codec preference in dial-peer:

Example:

voice class codec 1
     codec preference 1 g729r8
     codec preference 2 g711alaw
     codec preference 3 g711ulaw
dial-peer voice 70021 voip
     description Used for Switch leg SIP Direct
     preference 1
     max-conn 225 
     destination-pattern xxxx...... # Customer specific destination
     session protocol sipv2
     session target ipv4:###.###.###.### # IP Address for Unified CVP
     session transport tcp
     voice class codec 1
     voice-class sip options-keepalive up-interval 12 down-interval 65 retry 2
     dtmf-relay rtp-nte
     no vad
Step 2

Modify the dial-peer to specify the codec explicitly for a dial-peer: 

dial-peer voice 9 voip 
     description For Outbound Call for Customer
     destination-pattern <Customer Phone Number Pattern>
     session protocol sipv2
     session target ipv4:<Customer SIP Cloud IP Address>
     session transport tcp
     voice-class sip rel1xx supported "100rel"
     voice-class sip options-keepalive up-interval 12 down-interval 65 retry 2
     dtmf-relay rtp-nte
     codec g711alaw
     no vad

dial-peer voice 10 voip 
     description ***To CUCM Agent Extension For Outbound***
     destination-pattern <Agent Extension Pattern to CUCM>
     session protocol sipv2
     session target ipv4:<CUCM IP Address>
     voice-class sip rel1xx supported "100rel"
     dtmf-relay rtp-nte
     codec g711alaw

Configure VXML Gateway

Procedure

Modify the following dial-peer to specify the codec explicitly for a dial-peer: 

dial-peer voice 919191 voip
     description Unified CVP SIP ringtone dial-peer
     service ringtone
     incoming called-number 9191T
     voice-class sip rel1xx disable
     dtmf-relay rtp-nte
     codec g711alaw
     no vad

dial-peer voice 929292 voip
     description CVP SIP error dial-peer
     service cvperror
     incoming called-number 9292T
     voice-class sip rel1xx disable
     dtmf-relay rtp-nte
     codec g711alaw
     no vad

dial-peer voice 7777 voip
     description Used for VRU leg #Configure VXML leg where the incoming called
     service bootstrap
     incoming called-number 7777T
     dtmf-relay rtp-nte
     codec g711alaw
     no vad