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New Member

CUSP configuration

I have CUSP module in one of the 3845 router and i have configured it also, but i am not able to call the script. i am dialing from the phone registered to the callmanager. the system plays the message your calll cannot be completed as dialed.

the callmanager has a sip trunk to the gateway.

the callmanager has CVP also added as a trunk.

the gateway has a dial-peer sending the calls to the CUSP module IP.

Attached is the config of the CUSP module.

=====================================================

AW-IPCC-CUSP1(cusp-config)# show configuration ac
AW-IPCC-CUSP1(cusp-config)# show configuration active
Building CUSP configuration...
!
server-group sip global-load-balance call-id
server-group sip retry-after 0
server-group sip element-retries udp 2
server-group sip element-retries tls 1
server-group sip element-retries tcp 1
sip dns-srv
enable
no naptr
end dns
!
no sip header-compaction
no sip logging
!
sip max-forwards 70
sip network CVP standard
no non-invite-provisional
allow-connections
retransmit-count invite-client-transaction 3
retransmit-count invite-server-transaction 5
retransmit-count non-invite-client-transaction 3
retransmit-timer T1 500
retransmit-timer T2 4000
retransmit-timer T4 5000
retransmit-timer TU1 5000
retransmit-timer TU2 32000
retransmit-timer clientTn 64000
retransmit-timer serverTn 64000
udp max-datagram-size 1500
end network
!
sip network enterprise standard
no non-invite-provisional
allow-connections
retransmit-count invite-client-transaction 3
retransmit-count invite-server-transaction 5
retransmit-count non-invite-client-transaction 3
retransmit-timer T1 500
retransmit-timer T2 4000
retransmit-timer T4 5000
retransmit-timer TU1 5000
retransmit-timer TU2 32000
retransmit-timer clientTn 64000
retransmit-timer serverTn 64000
udp max-datagram-size 1500
end network
!
sip network service-provider standard
no non-invite-provisional
allow-connections
retransmit-count invite-client-transaction 3
retransmit-count invite-server-transaction 5
retransmit-count non-invite-client-transaction 3
retransmit-timer T1 500
retransmit-timer T2 4000
retransmit-timer T4 5000
retransmit-timer TU1 5000
retransmit-timer TU2 32000
retransmit-timer clientTn 64000
retransmit-timer serverTn 64000
udp max-datagram-size 1500
end network
!
sip overload reject retry-after 0
!
no sip peg-counting
!
sip privacy service
sip queue message
drop-policy head
low-threshold 80
size 2000
thread-count 20
end queue
!
sip queue radius
drop-policy head
low-threshold 80
size 2000
thread-count 20
end queue
!
sip queue request
drop-policy head
low-threshold 80
size 2000
thread-count 20
end queue
!
sip queue response
drop-policy head
low-threshold 80
size 2000
thread-count 20
end queue
!
sip queue st-callback
drop-policy head
low-threshold 80
size 2000
thread-count 10
end queue
!
sip queue timer
drop-policy none
low-threshold 80
size 2500
thread-count 8
end queue
!
sip queue xcl
drop-policy head
low-threshold 80
size 2000
thread-count 2
end queue
!
route recursion
!
sip tcp connection-timeout 30
sip tcp max-connections 256
!
no sip tls
!
trigger condition call-from-enterprise
sequence 1
  in-network enterprise
  end sequence
end trigger condition
!
trigger condition call-from-service-provider
sequence 1
  in-network service-provider
  end sequence
end trigger condition
!
trigger condition mid-dialog
sequence 1
  mid-dialog
  end sequence
end trigger condition
!
accounting
no enable
no client-side
no server-side
end accounting
!
server-group sip group cucm.ipcc.com enterprise
element ip-address 172.25.7.11 5060 udp q-value 1 weight 25
element ip-address 172.25.7.12 5060 udp q-value 1 weight 25
element ip-address 172.25.27.10 5060 udp q-value 1 weight 25
failover-resp-codes 503
lbtype weight
ping
end server-group
!
server-group sip group cvp.ipcc.com enterprise
element ip-address 172.25.27.165 5060 udp q-value 1 weight 25
failover-resp-codes 503
lbtype weight
ping
end server-group
!
server-group sip group gw.ipcc.com enterprise
element ip-address 172.25.14.106 5060 udp q-value 1 weight 25
failover-resp-codes 503
lbtype weight
ping
end server-group
!
policy lookup enterprise
sequence 1 enterprise request-uri uri-component user
  rule prefix
  end sequence
end policy
!
trigger routing sequence 1 by-pass condition mid-dialog
!
no server-group sip global-ping
!
sip cac session-timeout 720
sip cac enterprise 172.25.14.106 5060 udp limit -1
sip cac enterprise 172.25.27.10 5060 udp limit -1
sip cac enterprise 172.25.27.165 5060 udp limit -1
sip cac enterprise 172.25.7.11 5060 udp limit -1
sip cac enterprise 172.25.7.12 5060 udp limit -1
!
no sip cac
!
sip record-route enterprise udp 172.25.11.95 5060
sip listen enterprise udp 172.25.11.95 5060
!
call-rate-limit 30
!
end
AW-IPCC-CUSP1(cusp-config)#

route table enterprise
key 2 target-destination cucm.ipcc.com enterprise
key 77 target-destination cvp.ipcc.com enterprise
key 8001234567 target-destination gw.ipcc.com enterprise
key 88 target-destination cvp.ipcc.com enterprise
end route table

=====================================================

regards,

Sandeep

3 REPLIES
Green

CUSP configuration

This is a bit like saying you have just built a motor car but it's not working properly and you post a circuit diagram of the radio and ask us to help.

In the words of the inimitable Rodgers & Hammerstein:

Let's start at the very beginning

A very good place to start

When you read you begin with A-B-C

When you sing you begin with do-re-mi

When building up a CVP deployment, getting CUCM-originated calls working is the last piece of the puzzle. You should always start with calls into the gateway from either the PSTN (T1/E1) or an analogue phone into an FXS port, and work in stages, testing each stage, and adding complexity and redundancy until you reach the final setup.

Now if you don't have a T1/E1 or FXO for your gateway, and all you have is an IP phone, it's still advisable to build that the correct way, because you surely will have that PSTN connection eventually. I'll get to that part last.

This is the way I normally work through the CVP SIP build out.

(It appears that your NVRU label on CVP routing clients is 8001234567 and your agent extensions are similar to 2xxxx. )

Do

Call Server and no outbound proxy.

Gateway dial peer finds the Call Server through "session target ipv4::5060.

Call Server configured with "send to originator" to push the NVRU label 8001234567 back to the ingress gateway to start the VXML.

Add "send to originator" for the ringtone label 91919191

Add "send to originator" for the error label 92929292

Play a message, queue the call.

When an agent goes ready, a static route in the Call Server for 2> points to a subscriber

SIP trunk from CUCM to Call Server

Re

Remove the static route 2> from the Call Server

Change the Call Server to use the Outbound Proxy and restart the Call Server

Add a SIP trunk from CUCM to CUSP

Put your 2 pattern in CUSP to the Sub server group (your key 2 target-destination cucm.ipcc.com enterprise)

Mi

Change the dial peer in the gateway to send the call to the SIP Proxy

Add a route in the route table to find a Call Server (your key 77 and key 88 to the CVP server group)

Fa

Remove 8001234567, 91919191, 92929292 from the "send to originator" section

Add a route in the proxy for the NVRU label 8001234567 to find your gateway (your key 8001234567 for the gw server group)

Once you get this working, put the send to originators back for a branch-office deployment

So

IP phone calls a CTI Route Point mapped to the JTAPI user which has Dialed Number-Call Type-Scheduled Script

Add a label 8222222222 on the CUCM routing client to the NVRU

The first "Send to VRU" node returns 8222222222 to CUCM

CUCM has a Route Pattern on 8222* - the target is the SIP trunk to the Proxy server!

Add a route to the SIP proxy to find a Call Server for this pattern - in your case, add "key 82222 target-destination cvp.ipcc.com enterprise"

The script has a SECOND explicit "Send to VRU" to return the 8111111111 to CVP

CVP finds a gateway through the Proxy server - we set that up and tested it in the previous step

La

Check all the server groups in the Proxy are working correctly

Check that the OPTION PINGs are getting to all the elements and all server groups are up

Turn off/disable elements of the server groups in turn to ensure all redundancy is correctly set up (for the Call Servers, for the Subs, for the Gateways)

Ti

Configure a Server Group in the Call Servers for the 2 Proxy servers

Change the Call Server to use SRV local, no Outbound Proxy, and configure the target as the Server Group.

Add a second preference dial peer at the gateway to find the second CUSP or use srv records on the gateway

Do

Configure a Route Group/Route List of the two CUSP SIP Trunks in CUCM (probably Circular) and change the Route Pattern for 82222* to point to the Route List now instead of the trunk.

Remove the SIP trunk to the Call Server.

Now sing after me ...

Regards,

Geoff

New Member

CUSP configuration

Hi Geoff,

thanks for the detailed reply!

we are sending the 4 digit calls to the gateway via a sip trunk. Now that we have got the PRI i am translating the number to 4 digit number 8821.

the configuration looks the same as you mentioned and since we are not using the CTI route point i think the 'So' step is not required.

The "Ti" step is also not clear to me. i have created a sip server group. if i make no outbound proxy i dont have the option to select it in outbound proxy host and i cant put the server group name (FQDN) also.

If i send the call directly to the CVP without using Proxy it works fine but with the CUSP it doesnt work.

Regards,

Sandeep

New Member

CUSP configuration

Fantastic answer Geoff!  lol

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