Configuring a Gateway of Last Resort Using IP Commands

Commands ip default-gateway, ip default-network and ip route 0.0.0.0 0.0.0.0 explained:


http://www.cisco.com/en/US/tech/tk365/technologies_tech_note09186a0080094374.shtml

NX-OS and IOS Comparision

Cisco UCS servers

Название	Размер	CPU sockets	CPU	Память	Диски	I/O
UCS B200 M2	Половина	2	Intel Xeon 5600	12 DIMM 96 GB	2 SFF SAS	1 Mezz
UCS B230 M1	Половина	2	Intel Xeon 7500	32 DIMM 256 GB	2 SFF SAS	1 Mezz
UCS B250 M2	Целый	2	Intel Xeon 5600	48 DIMM 384 GB	2 SFF SAS	2 Mezz
UCS B440 M1	Целый	4	Intel Xeon 7500	32 DIMM 256GB	4 SFF SAS/SATA	2 Mezz
UCS C200 M2	1 RU	2	Intel Xeon 5600	12 DIMM 96 GB	4 SFF SAS/SATA	-
UCS C210 M2	2 RU	2	Intel Xeon 5600	12 DIMM 96 GB	16 SFF SAS/SATA	-
UCS C250 M2	2 RU	2	Intel Xeon 5600	48 DIMM 384 GB	4 SFF SAS/SATA	-
UCS C460 M1	4 RU	4	Intel Xeon 7500	64 DIMM 512 GB	12 SFF SAS/SATA	-

Cisco Expo-2010 Ukraine

Digit Manipulation

Digit Manipulation Techniques summary table:

Digit Stripping
POTS dial peers by default remove, or strip, any outbound digits that explicitly match their destination pattern.
To disable digit stripping use command:
 no digit-strip

Forward Digits
You can specify the exact number of digits to be forwarded. If the number of digits presented exceeds the number allowed, the rightmost
digits are sent.
forward-digits [number | all | extra]
where:
- number gives the number of digits to be forwarded.
- all means to forward all digits.
- extra tells the gateway to forward any digits that are longer than the length of the destination pattern.

Prefix Digits
You can transmit more than the dialed digits of a called number. Only for POTS dial peers.
prefix string

Number Expansion

Number expansion is another way to add digits to an outgoing called number; however, number expansion is applied to the gateway as a whole and manipulates only the called number. Number expansion manipulation occurs before any outbound dial peer is matched. Outbound dial peers match the expanded numbers, not the original ones.
num-exp original-number expanded-number

Caller ID
During an outgoing call, the CLID is sent as part of the call information. CLID information includes at least one calling party number. The CLID might also include a name, a second number, and redirecting number information.
CLID commands:

- clid network-number number Specifies the network number to be sent in the IE. It sets the presentation indicator to "Y" and the screening indicator to "network provided." Available in both dial-peer and voice service voip configuration modes.

- clid second-number strip Removes the original calling number from the H.225 source address field. You can also give this command on the same line as the clid network-number command. It is valid only if you have configured a network number.

- clid restrict Transmits the calling party information but sets the presentation indicator to "N" so that it is not displayed to the called party.

- clid strip [name | pi-restrict [all]] Removes the CLID number if just the clid strip command is given and sets the presentation indicator to "N." It removes the CLID name if the name option is added. To remove both the name and number, you must enter both commands separately. The pi-restrict option causes the CLID number to be stripped only when you set the progress indicator to "restricted." Adding the all keyword strips both the CLID number and name. The pi-restrict all option is available in both dial-peer and voice service voip configuration modes.

- clid substitute name Substitutes the calling number for the display name. Available in both dial-peer and voice service voip configuration modes.

Voice Translation Rules and Profiles

Using voice translation profiles for digit manipulation requires three steps:

Step 1.	Create one or more voice translation rules and a prioritized list of translations associated with each rule. A maximum of 128 rules is supported, with 15 translations per rule.
Step 2.	Create one or more voice translation profiles and associate the translation rules to the profile. You can define up to 1000 profiles, each with its own unique name. Within the profile, you can apply one voice translation rule to calling numbers, one to called numbers, and one to redirected called numbers.
Step 3.	Apply the voice translation profile to all VoIP calls globally, a dial peer, a voice port, a trunk group, a source IP group, or an interface.

Creating Voice Translation Rules

To create a voice translation rule, use the command voice translation-rule tag in global configuration mode. Then create an ordered list of one or more rules with the following command:

rule precedence /match pattern/ /replace pattern/ [type {match-type replace-type} [plan {match-type replace-type}]]

You can enter rules in any order; the precedence value determines the order in which the rules are executed. You can configure up to 15 rules.

Creating Voice Translation Profiles

(config)#voice translation-profile name
(cfg-translation-profile)#translate ?
  called           Translation rule for the called-number
  calling          Translation rule for the calling-number
  redirect-called  Translation rule for the redirect-number
(cfg-translation-profile)#translate called 1
(cfg-translation-profile)#translate calling 2
(cfg-translation-profile)#translate redirect-called 3

Applying Translation Profiles

After you create a voice translation profile, you can assign it to:

- Dial-Peer

- Voice Port

- Trunk Group

- VoIP Calls Globaly

- NFAS

- SRST

Blocking Calls

!
(config)#voice translation-rule 410
(cfg-translation-rule)#rule 1 reject /expression/
!
(config)#voice translation-profile BLOCK
(cfg-translation-profile)#translate calling 410
!
(config)#dial-peer voice 1 pots
(config-dial-peer)#call-block translation-profile incoming BLOCK
(config-dial-peer)#call-block disconnect-cause incoming call-reject
!

Order of Operation in Digit Manipulation

Translation Profile Order

 Troubleshooting Digit Manipulation

- test voice translation-rule rule-number phone-number Shows the results of a translation rule, enabling you to test it to ensure that it does what you planned
- debug voice translation Shows the translations happening
- show dialplan numbernumber Verifies number expansion and which dial peers a phone number matches
- debug voip ccapi inout Shows inbound and outbound dial peers being matched
- show num-exp[number] Displays the number expansion rules configured
- show dial-peer voice [tag] Displays any CLID, translation profiles, call blocking, disconnect cause, digit stripping, forwarding, or prefixing that is configured on the dial peer
- show voice translation-rule [number | sort [ascending|descending]] Lists the translation rules that are configured on the router and all translation patterns configured for each one
- show voice translation-profile [name | sort [ascending|descending]] Lists the translation profiles configured on the router and all translation rules associated with each one
- debug isdn q931 Shows the called and calling numbers sent out a PRI link for troubleshooting CLID commands
- csim start phone-number Simulates a phone call from the router; can be used with debugs

Cisco Unified Mobility

Consists:
- Mobile Connect: Allows an incoming call to a user's enterprise phone number to be offered to the user's office phone, as well as to up to ten configurable remote destinations. Typically such remote destinations are mobile phones and home office phones.
- Mobile Voice Access (MVA): similar to those of Mobile Connect for outgoing calls. Users who are outside the enterprise can make calls as if they were directly connected to CUCM. This functionality is commonly called
Direct Inward System Access (DISA) in traditional telephony environments.

Dial-peers Configuration

Dial-peer wildcard symbols:

 Examples of how these wildcard symbols are applied to the destination pattern and the dial string that results when dial string 4085551234 is matched to an outbound POTS dial peer.

 Matching Inbound Dial Peers

The router uses the full digit string received in the setup request for matching against the configured dial peers. The router or gateway matches call setup element parameters in the following order:
1. The router or gateway attempts to match the called number of the call setup request with the configured incoming called-number of each dial peer.
2. If a match is not found, the router or gateway attempts to match the calling number of the call setup request with the answer-address of each dial peer.
3. If a match is not found, the router or gateway attempts to match the calling number of the call setup request to the destination-pattern of each dial peer.
4. The voice port uses the voice port number associated with the incoming call setup request to match the inbound call leg to the configured dial peer port parameter.
5. If multiple dial peers have the same port configured, the router or gateway matches the first dial peer added to the configuration.
6. If a match is not found in the previous steps, dial peer 0 is matched.

Characteristics of the Default Dial Peer

Default dial peers are used for inbound matches only. Dial peer 0 has the following characteristics:
- Any codec
- IP precedence 0
- VAD enabled
- No RSVP support
- fax-rate service
- no ivr application

Matching Outbound Dial Peers

Outbound dial-peer matching is completed on a digit-by-digit basis. Therefore, the router or gateway checks for dial-peer matches after receiving each digit and then routes the call when a full match is made.
The router or gateway uses the dial peer destination-pattern command to determine how to route the call.

Device Mobility

When phones move between different CUCM sites, inaccurate phone settings may occur.
The phone configuration parameters that can be dynamically applied to the device configuration are grouped in two categories:
- Roaming-Sensitive Settings
 Date/Time Group (dp)
 Region (dp)
 Location (dp)
 Connection Monitor Duration (dp)
 Network Locale (dp, phone)
 SRST Reference (dp, phone)
 Media Resource Group List (dp, phone)
 Physical Location (dp)
 Device Mobility Group (dp)
- Device Mobility-Related Settings (impact on call routing)
 Device Mobility Calling Search Space (dp, phone)
 AAR Calling Search Space (dp, phone) (Calling Search Space only in the Phone Configuration not Line!)
 AAR Group (dp, phone)

Overlapping parameters configured at the phone have higher priority than settings at the home device pool and lower priority than settings at the roaming device pool.

Algorithm:
The current device pool is chosen as follows:
 - If the DMI is associated with the phone's home device pool, the phone is considered to be in its home location. Therefore, Device Mobility does not reconfigure the phone.
 - If the DMI is associated with one or more device pools other than the phone's home device pool, one of the associated device pools is chosen based on a round-robin load-sharing algorithm.
If the current device pool is different from the home device pool, the following checks are performed:
 - If the physical locations are not different, the phone's configuration is not modified.
 - If the physical locations are different, the roaming-sensitive parameters of the current roaming device pool are applied.
 - If the Device Mobility Groups are the same, in addition to different physical locations, the Device Mobility-related settings are also applied, along with the roaming-sensitive parameters.
When a user roams with a device from Germany to the U.S., all the roaming-sensitive settings are applied, but the Device Mobility-related settings are not applied. The phone now uses the PSTN gateway and dial rules of its home location even though the user moved to another site. The user does not have to adapt to the dial rules of the local site to which the phone was moved.


Configuration:
1. Configure physical locations.
2. Configure Device Mobility Groups.
3. Configure device pools.
4. Configure DMIs (that is, IP subnets).
5. Set the Device Mobility mode using the following:
 - A Cisco CallManager service parameter to set the default for all phones
 - The Phone Configuration window for an individual configuration for each phone.

Call Admission Control

CAC limits the number of calls between certain parts of the network to avoid bandwidth oversubscription, with too many voice calls over WAN links.

Locations:
The configured bandwidth limit is independent of the call's destination location. Unlike region configuration, in which the maximum permitted codec is configured for each pair of regions, the bandwidth limit of a location applies to all interlocation calls, regardless of the other location.

CUCM CAC is based on a hub-and-spoke topology, in which all calls over the WAN are monitored for CAC as if they go through HQ. Because the configured bandwidth limit does not consider the destination location, the 24-kbps limit of BR1 allows any call to go out or in, regardless of where it goes to or comes from. The headquarters limit is unaffected by such a call. Only locations BR1 and BR2 subtract 24 kbps from their limits. Because locations-based CAC does not provide topology awareness, CUCM does not even know that the call physically flows through headquarters. 

1. Add locations and configure the CAC bandwidth limit.
2. Assign locations to devices.

RSVP-Enabled Locations
RSVP can be enabled selectively between pairs of locations.
RSVP makes this CAC mechanism WAN topology-aware, because RSVP will communicate over the WAN. Standard locations do not contain in their configurations details of the WAN topology. RSVP-enabled locations work well with all topologies (full-mesh, partial-mesh, and hub-and-spoke) and adapt to network changes by considering the actual topology.
An RSVP agent is a device called a Media Termination Point (MTP) through which the call has to flow. RSVP is used only between the two RSVP agents. The Real-Time Transport Protocol (RTP) stream from the IP Phone to the RSVP agent does not use RSVP.
There are three separate RTP streams: Phonel talks to RSVP Agentl, RSVP Agentl talks to RSVP Agent2, and RSVP Agent2 talks to Phone2.
Media resource group lists are used to identify the RSVP agent to be used by an IP Phone.
Confiiguration:
1. Configure RSVP service parameters.
2. Configure RSVP agents in Cisco IOS.
!
dspfarm profile 1 mtp
 codec pass-through
 rsvp
 maximum sessions software 20
asswciate application SCCP
!
interface Serial0/0
 ip rsvp bandwidth 40
!
3. Add RSVP agents to CUCM.
 Cisco IOS Enhanced Software Media Termination Point
4. Enable RSVP between location pairs.
5. Configure media resource groups.
6. Configure media resource group lists.
7. Assign media resource group lists to devices.

AAR
AAR does not work with Survivable Remote Site Telephony (SRST). AAR is activated only after a call is denied by CAC, not by WAN failures.
AAR works only for calls placed to internal directory numbers. It does not apply to calls placed to route patterns or feature patterns such as Meet-Me or Call Park. However, it does work for hunt pilots and computer telephony interface (CTI) ports.
The alternative number used for the PSTN call is composed of the dialed directory number, a prefix configured per AAR source and destination group, and the external phone number mask of the called device.

Configuration
1. Configure AAR service parameters (Cisco CallManager service).
 - Automated Alternate Routing Enable
 - Out-of-Bandwidth Text
 - AAR Network Congestion Rerouting Text
2. Configure partitions and CCS.
3. Configure AAR groups. Assign a dial prefixes.
4. Configure phones for AAR:
 Phone Configuration:
 - AAR Calling Search Space
 - AAR Group (or directory number Group is used)
 Directory Number Configuration:
 - AAR Settings: Voice Mail checkbox
 - AAR Destination Mask (CFNB)
 - AAR Group
 - External Phone Number Mask

Multicast MOH from Branch Router Flash Configuration

All IP Phones must be able to access the main-site CUCM MOH server from their media resource group list.
Use Multicast for MOH Audio check box has to be checked at the media resource group that includes the multicast-enabled MOH server.
G.711 codec is used between the MOH server and the branch phones, because SRST multicast MOH supports only G.711.
Recommended practice dictates that you increment multicast on IP address instead of port number.

1. Enable multicast routing in the network.
 (config)#ip multicast-routing
 (config-if)#ip pim sparse-dense-mode
2. Configure multicast MOH in CUCM:
 a. Configure MOH audio sources for multicast MOH.
 b. Configure MOH audio server for multicast MOH.
 c. Configure the maximum hop value to prevent multicast MOH streams from being sent over the IP WAN.
3. Enable multicast MOH from branch router flash at the branch router.
call-manager-fallback
 ip source-address
 moh moh-file.au
 multicast moh 239.0.0.1 port 20120
4. Optional: Use alternative methods to prevent multicast MOH streams from being sent over the IP WAN:
 a. Use IP ACL at the IP WAN router interface.
 b. Disable multicast routing on the IP WAN router interface.