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SIP to carrier via MPLS with SRST H323 to POTS for backup

I have a customer running CM 4.1(3) with remote sites connected via MPLS using SIP (via CUBE) into MPLS cloud for PSTN connectivity. They also have 1-2 POTS lines to use for SRST if their T-1/MPLS/SIP connection goes down.

I've had a heck of a time getting the SRST/H323 dial-peers to work with the SIP dialpeer (and H323 dial-peer back to CUCM). Basically, I can't get SIP and SRST functioning simultaneously within the same CUBE/VGW. I've talked to TAC and have been clued into the use of the "monitor probe" dial-peer command with the call-fallback-active command to remove the SIP/primary-H323 dial-peers when the T-1/MPLS connection goes down; therefore, utilizing the SRST H323 dial-peers when in fallback mode. I have not been successful with this.

1) Does anyone have a working config that mimcs this environment, that they could share the highlights?

2) Does going to CUCM v7 help in any way, in determinng dial-peer selection? (Perhaps because I can then be SIP on both sides of the CUBE)?

Below are my config highlights:

!

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

h323

emptycapability

h225 id-passthru

h245 passthru tcsnonstd-passthru

sip

bind control source-interface Vlan241

bind media source-interface Vlan241

!

!

voice class h323 1

h225 timeout tcp establish 1

h225 timeout connect 60

h225 timeout setup 1

!

!

!

!

!

!

!

!

interface Serial0/2/0

ip address 2.2.2.2 255.255.255.0

encapsulation ppp

max-reserved-bandwidth 99

service-policy input WanIn

!

!

call fallback active

!

!

voice-port 0/0/0

timing hookflash-out 500

connection plar 8331

!

!

dial-peer voice 320 pots

*** TO CARRIER via POTS FOR H323/SRST ***

preference 8

destination-pattern 81[2-9]..[2-9]......

port 0/0/0

prefix 1

!

!

dial-peer voice 220 pots

*** TO CARRIER via POTS FOR H323/SRST ***

preference 8

destination-pattern 8[2-9]......

port 0/0/0

!

!

dial-peer voice 1 voip

*** TO CARRIER SIP ***

preference 1

destination-pattern .T

monitor probe icmp-ping 10.99.204.10

voice-class h323 1

session target ipv4:10.99.204.10

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer voice 2 voip

*** TO Subscriber FOR H323/SIP PRIMARY ***

preference 2

destination-pattern .T

monitor probe icmp-ping 10.1.1.2

session protocol sipv2

session target ipv4:10.1.1.2

incoming called-number .

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 3 voip

*** TO PUB FOR H323/SIP PRIMARY ***

preference 3

destination-pattern .T

monitor probe icmp-ping 10.1.1.1

voice-class h323 1

session target ipv4:10.1.1.1

dtmf-relay h245-alphanumeric

codec g711ulaw

no vad

!

dial-peer hunt 2

!

!

call-manager-fallback

secondary-dialtone 8

max-conferences 8 gain -6

limit-dn 7940 2

limit-dn 7960 6

timeouts interdigit 3

ip source-address 10.20.1.1 port 2000

max-ephones 42

max-dn 36

system message primary CM Failed, NOW on SRST

keepalive 10

!

Thanks for any input!

Mike.

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