Ive read numerous documents and books on h.323 & SIP CAC and even spoken to a CCIE the teaches CVOICE and im left a bit confused.
I read that you can enable CAC on a CME deployment using the call threshhold command and specify the maximum number of VoIP calls via an interface. Once this limit is reached, calls will be immediately re-routed to a pots dial-peer with higher preference.
But im told by the CCIE that this isnt possible. You can only specify a maximum number of calls through specific dial-peers.
Can someone please clarify?
Secondly, ive managed to get rtr responder with the ICPIF check, and call fallback works without problems. Its not ideal, but it works.
My question is, if the first CAC method works (max calls through and interface). Can it be combined with a measurement based mechanisim like ICPIF to double up on a CAC method?
Has anyobody got any examples that they have found to be really good with providing a reliable and quick failover to PSTN on CME without gatekeepers and RSVP?
Cisco IOS software has a built-in CAC mechanism with the call threshold interface command. This feature limits both inbound calls and outbound calls for a specific interface on the Unified CME router once a maximum threshold has been exceeded. For example, the following command causes calls from GigabitEthernet0/0 to be rejected after the number of simultaneous inbound/outbound calls exceeds five. Calls are then allowed once the maximum number of simultaneous calls falls below 3.
call threshold interface GigabitEthernet0/0 int-calls low 3 high 5
The benefit of this feature is that it does not require gatekeeping and can operate across multiple dial-peers. It is not subject to dial-peer maximum connection limitations. The limitation of this mechanism is that the maximum number of simultaneous calls is an aggregate of the total inbound and outbound calls. You cannot set up different thresholds for outbound or inbound calls using this mechanism.
Alternatively, you can use the VoIP Tandem Gateway feature of Cisco Unified CME 3.1 and above. This allows you to construct hub-and-spoke or hop-by-hop call routing arrangements. Hub-and-spoke call routing arrangements are historically common in small-scale voice over Frame Relay (VoFR) and voice over ATM (VoATM) networks. In these small-scale networks, you might have a single larger "hub" Cisco Unified CME system with approximately 100 users at a primary site, with perhaps five satellite Cisco Unified CME systems, each with 20 users linked on VoIP "spokes" to the primary. In this arrangement, only the central hub site needs VoIP dial peers to be configured to define the location of all network-wide extensions. The spoke satellite sites only need to know to send nonlocal calls to the hub site. The central hub site can then relay the call to the final spoke site destination.
Are you getting this error “Installer User Interface Mode Not Supported. The installer cannot run in this UI mode. To specify the interface mode, use the -i command-line option, followed by the UI mode identifier. The value UI mode identifiers...
The below trick might come handy when you have to add a new node to a cluster but you don't have or is unsure of the security password for the publisher. This procedure has been around for ages.
1) Login into the CLI of the Publisher.