Cisco Pre-Sales Question: Does the BE6000S come with Expressway

Answer: No, you'll have to host it separately on some other Server. You can however have it included in the BE6K-Medium or High.

Polycom IP6000/7000 Endpoint Registered to Cisco CUCM - Is the Polycom Endpoint able to use Cisco CUCM's Conference Resources to enable greater than 3 Party Conf. Calls (Ad-Hoc) or MeetMe

Question: Can the Polycom EndPoint IP6000/7000 use Centralized Conferencing Resources when registered to CUCM.

Answer: No

SoundStation IP phones only support three-way audio conference(local bridge) when they are deployed with CUCM.

 

Since SoundStation IP phones cannot use CUCM conferencing resources, we may need to initiate conference from Cisco IP phones when there are more than three participants in the conference.

Ref: Polycom TAC: 

SR#1-781454034 IP 6000 To using Cisco Call manager 10.5 for centralized control but no options found

Cisco Pre-Sales Question - Is the BE600S capable of being part of a Cisco Call Manager Cluster

Q: Can the BE6ks can be part of another cluster server?

A: "The quick answer is yes, BE6000S appliances may be clustered.

SRST allows call control resiliency to be accommodated within the "single box" solution.  The applications provided with the BE6000S support standard clustering with application instances on an additional server host, if that is required.  Note: Cluster capacity is limited to that of the smallest member (i.e. 150 users / 300 devices) where clustering between BE6000S and BE6000M or H hosts is considered."

Ref:  BE6ks (01304783) 

9971 SIP Phones fail to register to CME

Other than the below roughly make sure your SIP is bound to the same interface as your "source address" under "voice register global"... 

voice service voip

sip

bind control source-interface Loopback0

bind media source-interface Loopback0

** above the sip is bound to my interface Loopback0 and below the voice register global has the source-address mentioned as my voice vlan interface ip..... this was what prevented the 9971 phone from registering to CME, once I changed the sip bind to gi0/0.11 (same as the source-address... the phones registered)

voice service voip

    allow-connections h323 to h323

    allow-connections h323 to sip

    allow-connections sip to h323

    allow-connections sip to sip

sip

     bind control source-interface gi0/0.11 (was earlier set to Loopback0)

     bind media source-interface gi0/0.11 (was earlier set to Loopback0)

     registrar server expires max 300 min 60

voice register global

    max-dn 25

    max-pool 10

    mode cme

    source-address 142.102.66.254 port 5060

    load 9971 sip9971.9-4-1SR1-2

    authenticate register

    tftp-path flash:

    create profile

voice register dn  1

    number 3001

voice register pool  1

    id mac 5ca4.8a64.5e87

    type 9971

    number 1 dn 1

    username cisco password cisco

tftp-server flash:dkern9971.100609R2-9-4-1SR1-2.sebn

tftp-server flash:kern9971.9-4-1SR1-2.sebn

tftp-server flash:rootfs9971.9-4-1SR1-2.sebn

tftp-server flash:sboot9971.031610R1-9-4-1SR1-2.sebn

tftp-server flash:sip9971.9-4-1SR1-2.loads

tftp-server flash:skern9971.022809R2-9-4-1SR1-2.sebn

Calls from SIP 9971 phones to internal extensions takes too long to dial out (roughly 15 seconds)

adjust the t302 timer in CUCM from the Service Parameters (the lowest you can go as of CUCM version 9.x) is 3000 ms or 3 seconds.

Problem with AAR Calls - Remote H323 Gateway is not able to route the call to the CUCM

When AAR is triggered in CUCM (I mean once the phone says "Not Enough Bandwidth. Rerouting"... and attempts to dial the External Phone Number Mask entry of the destination phone, the call once routing over the PSTN with the E.164 reaches the far end router over PSTN but the return VOIP Dial Peer to CUCM does not seem to reach CUCM.

Fix:

"Bind your H323 traffic to a Voice Interface on the Remote Voice Gateway that is returning the call to CUCM and reregister the remote gateway in CUCM as H.323.



Also do the following (although the above H323 traffic binding fixed the problem)

voice service voip on both gateways and allowed-connections h to h

Also if it's a 15.0 IOS then either say "no IP Address Trusted Authenticate" or add the ip address of the source to the "IP Addresss Trusted List" under "voice service voip"

Overall UCS Connectivity between the VM's to the FI's

The above pic on shows one Blade Server on a Chassis connected to one side of the Fabric... the other side of the Fabric would have the same logic.