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Friday, December 17, 2010

SRST Dial Plan Voice Translation-Profile Commands for Digit Manipulation Best Cisco CCIE Training Institute in Delhi Gurgaon

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The combination of voice translation-profiles and voice translation-rules creates a very
powerful tool for modifying numbers so that they match dial plan needs. Example 6-12
shows the configuration commands for voice translation profiles.
Example 6-12 Voice Translation-Profile Commands
RemoteSite#configure terminal
R e m o t e S i t e ( c o n f i g ) # v o i c e t r a n s l a t i o n - p r o f i l e name
R e m o t e S i t e ( c f g - t r a n s l a t i o n - p r o f i l e ) # t r a n s l a t e { c a l l e d I c a l l i n g I
r e d i r e c t - c a l l e d I r e d i r e c t - t a r g e t } translation-rule-number
You define a translation profile for voice calls using the voice translation-profile command
in global configuration mode. The name parameter of this command defines the name of
the translation profile. The maximum length of the voice translation-profile name is 31
alphanumeric characters.
You associate a translation rule with a voice translation profile using the translate command
in voice translation-profile configuration mode. The following list defines the
keywords and parameter for the translate command:
• called associates the translation rule with called numbers.
• calling associates the translation rule with calling numbers.
• redirect-called associates the translation rule with redirected called numbers.
• redirect-target associates the translation rule with transfer-to numbers and callforwarding
final destination numbers.
• translation-rule-number is the number of the translation rule to use for the call
translation. The valid range is from 1 to 2147483647. There is no default value.
NOTE The prior IOS digit manipulation tool translation rule has been replaced by
voice translation-rule. The commands are similar but are incompatible with each other.
SRST Dial Plan Voice Translation-Profile Commands for Digit Manipulation 143
SRST Dial Plan Voice Translation-Rule Commands for
Number Modification
Example 6-13 shows the configuration commands for voice translation-rules.
Example 6-13 Voice Translation-Rule Commands
RemoteSite#configure terminal I
RemoteSite(config)#voice translation-rule number
r o u t e r ( c f g - t r a n s l a t i o n - r u l e ) # r u l e precedence /match-pattern/
/replace-pattern/[type {match-type replace-type} [plan
{match-type replace-type}))
You define a translation rule for voice calls using the voice translation-rule command in
global configuration mode. The number parameter identifies the translation rule. The range
of the number is from 1 to 2147483647. The choice of the number does not affect usage
priority.
You define a translation rule using the rule command in voice translation-rule configuration
mode. The following list defines the keywords and parameters for the rule command, as
shown in Example 6-13:
• The parameter precedence defines the priority of the translation rule. The range is from
1 to 15.
• The parameter /match-pattern/ is a stream editor (SED) expression used to match
incoming call information. The slash (/) is a delimiter in the pattern.
• The parameter/replace-pattern/ is a SED expression used to replace the match pattern
in the call information. The slash is a delimiter in the pattern.
• The optional construct type match-type replace-type lets you modify the call's number
type. Valid values for the match-type argument are abbreviated, any, international,
national, network, reserved, subscriber, and unknown. Valid values for the replacetype
argument are abbreviated, international, national, network, reserved,
subscriber, and unknown.
• The optional construct plan match-type replace-type lets you modify the call's
numbering plan. Valid values for the match-type argument are any, data, ermes, isdn,
national, private, reserved, telex, and unknown. Valid values for the replace-type
argument are data, ermes, isdn, national, private, reserved, telex, and unknown.
144 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
SRST Dial Plan Profile Activation Commands for Number Modification
Voice translation profiles can be bound to dial peers, source groups, trunk groups, voice
ports, and the voice service POTS.
Example 6-14 shows the configuration commands for voice translation profile activation.
Example 6-14 Voice Translation Rule Activation Commands
RemoteSite#configure terminal
RemoteSite(config)#voice-port 0/1/0:23
RemoteSite(config-voiceport) # t r a n s l a t i o n - p r o f i l e {incoming I outgoing} name
R e m o t e S i t e ( c o n f i g - v o i c e p o r t ) # e x i t
RemoteSite(config)#call-manager-fallback
r o u t e r ( c o n f i g - c m - f a l l b a c k ) # t r a n s l a t i o n - p r o f i l e {incoming I outgoing} name
In this example, the voice translationprofile is bound to a voice port. The voice
translationprofile can also be bound to all the dial peers, but the voice port needs to
be done only once.
You assign a translation profile to a voice port using the translation-profile command in
voice-port configuration mode. The following list defines the keywords and parameter for
the translation-profile command:
• The keyword incoming specifies that this translation profile handles incoming calls.
• The keyword outgoing specifies that this translation profile handles outgoing calls.
• The parameter name is the name of the translation profile.
In addition to the configuration shown in Example 6-14, the voice translation profiles can
be bound to the call-manager-fallback Cisco IOS service. The structure of the command
is identical.
NOTE The incoming direction of the voice translation-profile bound to the callmanager-
fallback Cisco IOS service handles the calls coming from IP Phones that are
registered with the router.
For more information about voice translation profiles, refer to the following documents at
Cisco.com, which you should be able to locate by searching by title:
• TechNotes Number Translation Using Voice Translation Profiles
• TechNotes Voice Translation Rules
SRST Dial Plan Class of Restriction Commands 145
SRST Dial Plan Class of Restriction Commands
Calling privileges can be assigned to IP Phones when they are in SRST mode using COR
commands. In the absence of COR in SRST dial peers, all phones can dial all numbers.
Example 6-15 shows the dial plan configuration commands for COR as they apply to SRST.
Example 6-15 Class of Restriction Commands
RemoteSite#configure terminal
RemoteSite(config)#call-manager-fallback
RemoteSite(config-cm-fallback)#cor {incoming I outgoing} cor-list-name
[cor-list-number starting-number - ending-number I default]
The command cor configures a COR on dial peers that are associated with directory
numbers. The following list defines the keywords and parameters for the cor command:
• The keyword incoming defines that a COR list is to be used by incoming dial peers.
• The keyword outgoing defines that a COR list is to be used by outgoing dial peers.
• The parameter cor-list-name is the COR list name.
• The parameter cor-list-number is a COR list identifier. The maximum number of COR
lists that can be created is 20, composed of incoming or outgoing dial peers. The first
six COR lists are applied to a range of directory numbers. The directory numbers that
do not have a COR configuration are assigned to the default COR list, as long as a
default COR list has been defined.
• The parameters starting-number - ending-number define the directory number range,
such as 2000 to 2025.
• The keyword default instructs the router to use an existing default COR list.
Table 6-2 summarizes the functions of COR dialed calls.
146 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
Table 6-2 COR Dialing Possibilities
COR List on Incoming Dial
Peer
COR List on Outgoing Dial
Peer Result
No COR No COR The call succeeds.
No COR A COR list is applied for
outgoing calls.
The call succeeds. By default,
the incoming dial peer has the
highest COR priority when no
COR is applied. If you apply no
COR for an incoming call leg
to a dial peer, the dial peer can
make a call out of any other dial
peer, regardless of the COR
configuration on the outgoing
dial peer.
A COR list is applied for
incoming calls.
No COR The call succeeds. By default,
the outgoing dial peer has the
lowest priority. Because some
COR configurations exist for
incoming calls on the incoming
or originating dial peer, it is a
superset of the outgoing-call
COR configuration for the
outgoing or terminating dial
peer.
A COR list is applied for
incoming calls (a superset of
the COR list applied for outgoing
calls on the outgoing dial
peer).
A COR list is applied for
outgoing calls (subsets of the
COR list applied for incoming
calls on the incoming dial peer).
The call succeeds. The COR
list for incoming calls on the
incoming dial peer is a superset
of the COR list for outgoing
calls on the outgoing dial peer.
A COR list is applied for
incoming calls (a subset of the
COR list applied for outgoing
calls on the outgoing dial peer).
A COR list is applied for
outgoing calls (supersets of the
COR list applied for incoming
calls on the incoming dial peer).
The call does not succeed. The
COR list for incoming calls on
the incoming dial peer is not a
superset of the COR list for
outgoing calls on the outgoing
dial peer.
NOTE The complete configuration of COR is handled in the Cisco Voice over IP
course. Table 6-2 presents an overview only.
SRST Dial Plan Example
Figure 6-8 shows a multisite topology with a Cisco Unified SRST-enabled Cisco IOS router
in the remote site.
SRST Dial Plan Class of Restriction Commands 147
Figure 6-8 SRST Dial Plan Topology
Main Site
Cisco Unified
Communications
Manager
Phonel Phone2 Phone3
This figure shows a main site with a PSTN number of 51 l-555-2xxx and a remote site with
a PSTN number of 521-555-3xxx. Four digits are used for all internal calls, including calls
between the main site and remote site. The remote-site gateway has a single ISDN PRI
connection to the PSTN configured on port 0/1/0:23.
For the SRST remote-site configuration shown in Example 6-16, assume that the remote
site has only three phones, with one DN each. During SRST fallback, Phone 1 is configured
with directory number 3001 and has unlimited PSTN dialing access. Phone 2 is configured
with directory number 3002 and is not be allowed to place international calls. Phone 3 is
configured with directory number 3003 and is allowed to place only internal calls. Fourdigit
dialing to headquarters is configured, and calls should be sent to the main site over the
PSTN when in SRST mode.
The remote-site router also requires MGCP configurations, as discussed previously, but
they are not included in Example 6-16 for simplicity.
Example 6-16 Remote-Site SRST Dial Plan Configuration Example
a p p l i c a t i on
g l o b a l
s e r v i c e a l t e r n a t e d e f a u lt
i

c a l l - m a n a g e r - f a l l b a ck
ip source-address 10.1.250.101 port 2000
max-ephones 3
max-dn 3
cor incoming phonel 1 3001
cor incoming phone2 2 3002
continues
148 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
Example 6-16 Remote-Site SRST Dial Plan Configuration Example (Continued)
cor incoming phone3 3 3003
d i a l p l a n - p a t t e r n 1 5215553...
e x t e n s i o n - l e n g t h 4
I
d i a l - p e e r cor custom
name i n t e r n al
name pstn
name p s t n - i n tl
I
d i a l - p e e r cor l i s t i n t e r n al
member i n t e r n al
!
d i a l - p e e r cor l i s t pstn
member pstn
I
d i a l - p e e r cor l i s t p s t n - i n tl
member p s t n - i n tl
I
d i a l - p e e r cor l i s t phonel
member i n t e r n al
member pstn
member p s t n - i n tl
I
d i a l - p e e r cor l i s t phone2
member i n t e r n al
member pstn
I
d i a l - p e e r cor l i s t phone3
member i n t e r n al
I
d i a l - p e e r voice 1 pots
d e s c r i p t i o n I n t e r n a l d i a l i n g from the PSTN
incoming called-number .
d i r e c t - i n w a r d - d i a l
p o r t 0/1/0:23
!
d i a l - p e e r voice 9 pots
d e s c r i p t i o n PSTN d i a l 9 f i r s t
c o r l i s t outgoing pstn
d e s t i n a t i o n - p a t t e r n 9T
port 0/1/0:23
!
d i a l - p e e r voice 9011 pots
d e s c r i p t i o n I n t e r n a t i o n a l d i a l 9 f i r s t
c o r l i s t outgoing p s t n - i n tl
SRST Dial Plan Class of Restriction Comman ds 149
Example 6-16 Remote-Site SRST Dial Plan Configuration Example (Continued)
d e s t i n a t i o n - p a t t e r n 9011T
p o r t 0/1/0:23
p r e f i x 011
I

d i a l - p e e r voice 2000 pots
d e s c r i p t i o n I n t e r n a l 4 d i g i t d i a l i n g to HQ
c o r l i s t outgoing i n t e r n al
t r a n s l a t i o n - p r o f i l e outgoing to-HQ
d e s t i n a t i o n - p a t t e r n 2 . ..
port 0/1/0:23
I
v o i c e t r a n s l a t i o n - r u l e 1
r u l e 1 r2l /15115552/
i

v o i c e t r a n s l a t i o n - p r o f i l e to-HQ
t r a n s l a t e c a l l e d 1
The first part of the SRST configuration includes the dialplan-pattern command
configured under call-manager-fallback configuration mode, which maps the internal
four-digit directory numbers to the E.164 PSTN number.
COR lists are configured for internal destinations called internal, for international PSTN
destinations named pstn-intl, and for all other PSTN destinations labeled pstn. These COR
lists are applied to dial peers as outgoing COR lists. Their function is equivalent to partitions
in CUCM.
Additional COR lists are configured, one per phone. These are applied as incoming COR
lists to phone directory numbers using the cor incoming command in call-managerfallback
configuration mode. The configuration shown in Example 6-16 applies the
incoming COR list phonel, which is equivalent to CSSs in CUCM, to phone 1, which
registers with the SRST gateway with a directory number of 3001, incoming COR list
phone2 to the phone with directory number 3002, and incoming COR list phone3 to the
phone with directory number 3003.
Outgoing COR lists are applied to the dial peers that are used as outgoing dial peers: dial
peer 9011 for international PSTN calls, dial peer 9 for PSTN calls, and dial peer 2000 for
calls to headquarters.
150 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
NOTE For simplicity, Example 6-16 does not show all the outbound dial peers, as
shown previously in Example 6-9. Using the destination pattern 9T as shown in dial
peer 9 typically is avoided when possible for local or national calls to avoid the interdigit
timeout associated with the T wildcard.
Dial peer 1 is configured for inbound dialing from the PSTN with the incoming callednumber
command to identify all destination phone numbers. Direct inward dialing is enabled,
which turns off the second dial tone at ISDN port 0/1/0:23 for external calls dialing in.
The called E.164 numbers (521-555-3xxx) are mapped to four-digit extensions because of
the dialplan-pattern command that is configured in call-manager-fallback configuration
mode. As a result, incoming PSTN calls are sent to the four-digit extensions.
Outgoing calls to phones located at the main site at extensions 2xxx match a destination
pattern in dial peer 2000. Dial peer 2000 sends calls to port 0/1/0:23 after performing digit
manipulation using the to-HQ voice translation profile. This profile translates the four-digit
called number to an 11-digit E. 164 PSTN number. The result is that during SRST fallback,
users can still dial four-digit extensions to reach phones in headquarters.

SRST Dial Plan of CFUR and CSS Best Cisco CCIE Security Training Center in delhi gurgaon

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The CFUR feature is a way to reroute calls placed to a temporarily unregistered destination
phone. The configuration of CFUR consists of the two main elements of destination
selection and CSS, as shown in Figure 6-6. Choose Cisco Unified Communications
Manager Administration > Call Routing > Directory Number.
Figure 6-6 SRST Dial Plan Configuration of CFUR and CSS
When the directory number is unregistered, calls can be rerouted to the voice mail that
is associated with the extension or to a directory number that is used to reach the phone
through the PSTN. The latter approach is preferable when a phone is located within a site
whose WAN link is down. If the site is equipped with SRST, the phone (and its co-located
PSTN gateway) reregisters with the co-located SRST router. The phone then can receive
calls placed to its PSTN direct inward dialing (DID) number.
134 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
In this case, the appropriate CFUR destination is the corresponding PSTN DID number of
the original destination directory number. Configure this PSTN DID in the destination field,
along with applicable access codes and prefixes. For this example, the number would be
9-1-481-555-0001.
CUCM attempts to route the call to the configured destination number using the CFUR CSS
of the called directory number. The CFUR CSS is configured on the target phone and is
used by all devices that are calling the unregistered phone. This means that all calling
devices use the same combination of route pattern, route list, route group, and gateway to
place the call, and that all CFUR calls to a given unregistered device are routed through the
same unique gateway, regardless of where the calling phone is located. It is recommended
that you select a centralized gateway as the egress point to the PSTN for CFUR calls. You
also should configure the CFUR CSS to route calls that are intended for the CFUR
destination to this centralized gateway.
SRST Dial Plan: Max Forward UnRegistered
Hops to DN
The CUCM service parameter Max Forward UnRegistered Hops to DN reduces the
impact caused by CFUR routing loops, as shown in Figure 6-7. Choose CUCM
Administration > System > Service Parameter > Cisco CallManager.
Figure 6-7 SRST Dial Plan Configuration of Max Forward UnRegistered Hops to DN
This parameter specifies the maximum number of forward unregistered hops that are
allowed for a directory number at one time. It limits the number of times the call can be
forwarded because of the unregistered directory number when a forwarding loop occurs.
Use this count to stop forward loops for external calls that have been forwarded by CFUR,
SRST Dial Plan Components for Normal Mode Analogy 135
such as intercluster IP Phone calls and IP Phone-to-PSTN phone calls that are forwarded to
each other. CUCM terminates the call when the value that is specified in this parameter is
exceeded. The default 0 disables the counter but not the CFUR feature. The allowed range
is from 0 to 60.
MGCP Fallback and SRST Dial Plan Configuration
in the Cisco IOS Gateway
A dial plan in SRST mode, at a minimum, enables the remote-site users to place and receive
calls from the PSTN.
At least one dial peer needs to be configured to enable calls to and from the PSTN. The
destination pattern of that dial peer has to correspond to the PSTN access code (for example,
9T). The more elegant way is to configure several dedicated dial peers with destination
patterns that match the number patterns in a closed numbering plan, such as 91
(91 followed by ten dots).
In countries that have variable dial plans, the only destination pattern that is needed is 9T.
Because of the variable length of dialed numbers, the router waits for the interdigit timeout
(T302) or for a hash (#) sign to indicate the end of the dial string. Cisco Unified SRST
version 4.1 and Cisco Unified Communications Manager Express Release 4.1 do not
support the overlap sending feature to the PSTN. The receiving of ISDN overlap dialing
from PSTN is supported but has to be enabled on the interfaces. To shorten the wait time
for users after they complete the dial string, it is possible to reduce the interdigit timeout
from the SRST default of 10 seconds.
SRST Dial Plan Components for Normal
Mode Analogy
A good SRST dial plan is as close as possible between the dialing functionality in normal
mode and in SRST mode. The telephony service should have the same look and feel for the
user, regardless of the mode the system is in. For example, it would be an unacceptable
failover design if the remote user required an awareness of WAN link connectivity when he
dials his headquarters.
The numbers in the call lists (such as missed calls) must have the correct format (PSTN
access code plus PSTN phone number) to enable users to use the list entries for dialing. The
calling party ID of incoming calls from the PSTN needs to be modified by voice translation
profiles and voice translation rules.
136 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
Abbreviated dialing between sites of the site code plus the extension number is possible in
SRST mode. Voice translation profiles have to be used to expand the called numbers to
PSTN format for intersite dialing.
If the calling privileges (which normally are controlled by the CUCM) have to be preserved
in SRST mode, class of restriction (COR) configuration must be used.
SRST Dial Plan Dial Peer Commands
The dial-peer command is the main component for configuring dial plans on Cisco IOS
routers, as shown in Example 6-8.
Example 6-8 Dial Peer Commands for an SRST Dial Plan
RemoteSite#configure terminal
RemoteSite(config)#dial-peer voice tag pots
RemoteSite(config-dial-peer)#destination-pattern [+]string[T]
R e m o t e S i t e ( c o n f i g - d i a l - p e e r )#port slot-number/port
You define a particular dial peer, specify the voice encapsulation method, and enter dial
peer configuration mode using the dial-peer voice command in global configuration mode.
The following list defines the keywords and parameters used in the configuration shown in
Example 6-8:
• The parameter tag specifies digits that define a particular dial peer. The range is from
1 to 2147483647.
• The keyword pots indicates that this is a plain old telephone service (POTS) peer. The
option voip also exists, indicating that this is a VoIP peer, but is not mentioned in
Example 6-8 because POTS dial peers are predominately used for SRST. POTS dial
peers contain a port, whereas VoIP dial peers contain a configured IP address.
• You specify either the prefix or the full E.164 telephone number to be used for a dial
peer using the destination-pattern command in dial peer configuration mode.
• The optional character + indicates that an E. 164 standard number follows.
SRST Dial Plan Dial Peer Commands 137
• The parameter string defines a series of digits that specify a pattern for the E. 164 or
private dialing plan telephone number. Valid entries are the digits 0 through 9, the
letters A through D, and the following special characters:
*
#
+
A
S
\
[]
0
• The optional control character T indicates that the destination-pattern value is a
variable-length dial string. Using this control character enables the router to wait until
all digits are received before routing the call.
• To associate a dial peer with a specific voice port, use the port command in dial peer
configuration mode.
• The parameter slot-number defines the number of the slot in the router in which the
voice interface card (VIC) is installed. Valid entries depend on the number of slots that
the router platform has.
• The parameter port defines the voice port number. Valid entries are 0 and 1.
Table 6-1 lists common classes of PSTN calls in the North American Numbering Plan
(NANP) and lists the pattern that is used for each class. An access code of 9 should be used
to indicate a PSTN call. The exception is 911, which should be configured with and without
the access code 9. The patterns outlined in Table 6-1 must be reachable in SRST mode.
138 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
An access code of 9 typically is used to indicate a PSTN call; however, other access codes
such as 8 are also permissible.
The patterns in the table are the minimum number of patterns that need to be reachable in
SRST mode.
Example 6-9 provides a sample configuration of dial peers only, demonstrating outbound
dialing to the PSTN in an SRST router. This example is in the NANP, which also shows
local ten-digit dialing in area code 919. The configuration as written can be directly pasted
into a router.
Example 6-9 Dial Peer Example for an SRST Dial Plan
SRST Dial Plan Dial Peer Comman ds 139
NOTE This example does not contain any class of restriction (COR), which currently
allows all SRST registered phones to dial all numbers, including long-distance, 900, and
international, without any constraint. COR is discussed later in this chapter. In addition,
dial peers 3 and 4 do not require the forward-digits command, and are added only for
clarity.
140 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
SRST Dial Plan Commands: Open Numbering Plans
Example 6-10 defines the configuration commands for open numbering plans in an SRST
dial plan.
Example 6-10 SRST Dial Plan Commands for an SRST Dial Plan
RemoteSite#configure terminal
RemoteSite(config)#interface s e r i a l 0 / 1 / 0 : 23
R e m o t e S i t e ( c o n f i g - i f ) # i s d n overlap-receiving [T302 ms]
R e m o t e S i t e ( c o n f i g - i f ) # e x i t
RemoteSite(config)#call-manager-fallback
RemoteSite(config-cm-fallback)#timeouts i n t e r d i g i t sec
RemoteSite(config- cm - f a l l b a c k)#dialplan-pattern tag pattern
extension-length length [extension-pattern extension-pattern] [no-reg]
The following list defines the commands, keywords, and parameters used in the
configuration shown in Example 6-10:
• To activate overlap receiving on ISDN interfaces, you use the isdn overlap-receiving
command in interface configuration mode. This command is applicable on BRI interfaces
or on the ISDN interface of El/Tl controllers in PRI mode.
• The optional parameter T302 defines how many milliseconds the T302 timer should
wait before expiring. Valid values for the ms argument range from 500 to 20000. The
default value is 10000 (10 seconds).
CAUTION Modifying the T302 parameter, when connected to public networks, might
disable the function. The T302 describes the interdigit timeout for all phones in the
CUCM cluster.
• Configure the timeout value to wait between dialed digits for all Cisco IP Phones that
are attached to a router using the timeouts interdigit command in call-managerfallback
configuration mode.
• The parameter sec defines the interdigit timeout duration, in seconds, for all Cisco IP
Phones. Valid entries are integers from 2 to 120.
• Create a global prefix that can be used to expand the extension numbers of inbound and
outbound calls into fully qualified E.164 numbers using the dialplan-pattern
command in call-manager-fallback configuration mode.
• The parameter tag is the unique identifier that is used before the telephone number. The
tag number is from 1 to 5.
SRST Dial Plan Commands: Open Numbering Plans 141
• The parameter pattern is the dial plan pattern, such as the area code, the prefix, and the
first one or two digits of the extension number, plus wildcard markers or dots (.) for the
remainder of the extension-number digits.
• The keyword extension-length sets the number of extension digits that will appear as
a caller ID followed by the parameter length, which is the number of extension digits.
The extension length must match the setting for IP Phones in CUCM mode. The range
is from 1 to 32.
• The optional keyword extension-pattern sets the leading digit pattern of an extension
number when the pattern is different from the leading digits defined in the pattern
variable of the E.164 telephone number. An example is when site codes are used. The
parameter extension-pattern that follows defines the leading digit pattern of the extension
number. It is composed of one or more digits and wildcard markers or dots (.)• For
example, 5,. would include extensions 500 to 599, and 5... would include extensions
5000 to 5999. The extension pattern configuration should match the mapping of internal
to external numbers in CUCM.
• The optional keyword no-reg prevents the E. 164 numbers in the dial peer from registering
with the gatekeeper.
Example 6-11 demonstrates the use of the dialplan-pattern command, which shows how
to create a dial plan pattern for directory numbers 500 to 599 that is mapped to a DID range
of 408-555-5000 to 5099. If the router receives an inbound call to 408-555-5044, the dial
plan pattern command is matched, and the extension of the called E. 164 number, 408-555-
5000, is changed to directory number 544. If an outbound calling party extension number
(544) matches the dial plan pattern, the calling-party extension is converted to the E.164
number 408-555-5044. The E.164 calling-party number appears as the caller ID.
Example 6-11 SRST Dial Plan Example for Mapping Directory Numbers
RemoteSite#configure terminal I
RemoteSite(config)#call-manager-fallback
RemoteSite(config-cm-fallback)#dialplan-pattern 1 40855550..
extension-length 3 extension-pattern 5..

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This section describes the configuration to adjust the CUCM dial plan to work with Cisco
Unified SRST.
The CUCM dial plan has to be adjusted to ensure the reachability of remote-site phones by
their extensions even if the remote site runs in SRST mode. The parameter that enables this
adjustment is the CFUR destination setting, which has to be defined on every line of an
SRST enabled remote-site phone. This parameter was introduced in Cisco Unified Communication
Manager Release 4.2.
The CFUR feature forwards calls to unregistered (disconnected or logged out) directory
numbers for the defined destination. The destination might be the PSTN number of a phone
at a remote site or the voice mail for a user in a CUCM Extension Mobility setting.
SRST Dial Plan of CFUR and CSS 133
To ensure that the feature works even if a major WAN breakdown disconnects all the remote
sites, only voice gateways located at the main site should be used. This can be ensured by
selecting the correct Calling Search Space (CSS) for the CFUR destination.
CFUR causes routing loops whenever there is a single disconnected SRST phone in
which the remote location is not in SRST mode. Internal calls to that directory number
are forwarded to the CFUR (PSTN) destination and are received by the remote-site gateway
in normal mode. This gateway handles the call as usual, sending the signaling to its
CUCM subscriber. CUCM then again forwards the call to the PSTN, causing an inevitable
routing loop.
To limit the impact of these routing loops, Cisco introduced a CUCM service parameter
called Max Forward UnRegistered Hops to DN. When activated, this counter limits the
calls that are forwarded to one CFUR destination.

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To configure the MGCP gateway fallback on a Cisco IOS router to support the MGCP
fallback function, follow these steps:
Step 1 Activate MGCP gateway fallback.
Step 2 Define the service to fall back to.
To enable outbound calls while in SRST mode on an MGCP gateway, you must configure
two fallback commands on the MGCP gateway. These two commands allow SRST to
assume control over the voice port and over call processing on the MGCP gateway. With
Cisco IOS Software releases before 12.3(14)T, configuring MGCP gateway fallback involves
the ccm-manager fallback-mgcp and call application alternate commands. With Cisco
IOS Software releases after 12.3(14)T, configuring MGCP gateway fallback uses the
ccm-manager fallback-mgcp and service commands.
NOTE Both commands have to be configured. Configurations will not work reliably if
only the ccm-manager fallback-mgcp command is configured.
To use SRST on an MGCP gateway, you must configure SRST and MGCP gateway
fallback on the same gateway.
Example 6-4 Cisco Unified SRST Configuration Example
IRemoteSite#configure terminal
RemoteSite(config)#call-manager-fallback
RemoteSite(config-cm-fallback)#ip source-address 172.47.2.1 port 2000
RemoteSite(config-cm-fallback)#max-ephones 3 dual-line
RemoteSite(config-cm-fallback)#max-dn 6
RemoteSite(config-cm-fallback)#limit-dn 7960 2
RemoteSite(config-cm-fallback)#keepalive 20
RemoteSite(config-cm-fallback)#end
RemoteSite#
MGCP-Gateway-Fallback Configuration on the Cisco IOS Gateway 131
MGCP Fallback Activation Commands
The Cisco IOS command ccm-manager fallback-mgcp, shown in Example 6-5, enables
the gateway fallback feature and allows an MGCP voice gateway to provide call-processing
services through SRST or other configured applications when CUCM is unavailable.
Example 6-5 MGCP Fallback Activation Commands
RemoteSite#configure terminal
RemoteSite(config)#ccm-manager fallback-mgcp
RemoteSite(config) # c a l l application alternate Default
RemoteSite(config-app-global)#service alternate Default
The call application alternate Default command specifies that the default voice application
takes over if the MGCP call agent is unavailable. This allows a fallback to H.323 or
SIP, which means that local dial peers are considered for call routing.
The service alternate Default command is entered in the global-configuration submode of
the application-configuration submode. To navigate to this location, follow these steps:
Step 1 To enter application configuration mode to configure applications, use
the application command in global configuration mode.
Step 2 To enter application-configuration global mode, use the global command
in application configuration mode.
Enter either of the two commands, depending on the Cisco IOS Software release. The
newer configuration method is the service command.
As discussed in the preceding chapter, analog calls are preserved in the event of MGCP
fallback. To provide call preservation during switchback, call preservation for H.323 has to
be enabled using the commands shown in Example 6-6.
Example 6-6 H.323 Call Preservation Activation Commands
RemoteSite#configure terminal
RemoteSite(config)#voice service voip
RemoteSite(conf-voi-serv)#h323
RemoteSite(conf-serv-h323)#no h225 timeout keepalive
MGCP Fallback Configuration Example
Figure 6-5 shows an MGCP-controlled remote-site gateway with an MGCP-gatewayfallback
configuration for an SRST-enabled Cisco IOS router, as shown in Example 6-7.
132 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
Figure 6-5 MGCP Fallback Example
Main Site
Example 6-7 MGCP Fallback Configuration Example
RemoteSite#configure terminal
RemoteSite(config)#ccm-manager fallback-mgcp
RemoteSite(config)#application
RemoteSite(config-app)#global
RemoteSite(config-app-global)#service alternate Default
RemoteSite(config-app- global)#end
RemoteSite#
NOTE More commands might be necessary, depending on the complexity of the
deployment.

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To configure Cisco Unified SRST on a Cisco IOS router to support the Cisco IP Phone
functions, follow these steps:
Step 1 Enter call-manager-fallback configuration mode to activate SRST.
Step 2 Define the IP address and port to which the SRST service binds.
Step 3 Define the maximum number of directory numbers to support.
Step 4 Define the maximum number of IP Phones to support.
Step 5 Define the maximum number of numbers allowed per phone type.
Step 6 Define the phone keepalive interval.
TIP When Cisco Unified SRST is enabled, Cisco IP Phones in call-manager-fallback
configuration mode do not have to be reconfigured, because phones retain the same
configuration that was used with CUCM.
SRST Configuration on the Cisco IOS Gateway 127
SRST Activation Commands
Example 6-1 shows the commands for the first two SRST configuration steps.
Example 6-1 SRST Activation Commands
RemoteSite#configure terminal
RemoteSite(config)#call-manager-fallback
R e m o t e $ i t e ( c o n f i g - c m - f a l l b a c k ) # i p source-address ip-address [port port]
[any-match I strict-match]
The Cisco IOS command call-manager-fallback enters call-manager-fallback configuration
mode.
The Cisco IOS command ip source-address enables the router to receive messages from
the Cisco IP Phones through the specified IP addresses and provides for strict IP address
verification. The default port number is 2000. This IP address will be supplied later as an
SRST reference IP address in CUCM Administration.
The ip source-address command is mandatory. The fallback subsystem does not start if the
IP address of the Ethernet port to which the IP Phones are connected (typically the Ethernet
interface of the local SRST gateway) is not provided. If the port number is not provided,
the default value (2000) is used.
The any-match keyword instructs the router to permit Cisco IP Phone registration even
when the IP server address used by the phone does not match the IP source address. This
option lets you register Cisco IP Phones on different subnets or those with different default
DHCP routers or different TFTP server addresses.
The strict-match keyword instructs the router to reject Cisco IP Phone registration attempts
if the IP server address used by the phone does not exactly match the source address. By
dividing the Cisco IP Phones into groups on different subnets and giving each group different
DHCP default router or TFTP server addresses, this option restricts the number of Cisco
IP Phones allowed to register.
SRST Phone Definition Commands
The commands shown in Example 6-2, max-dn and max-ephones, are mandatory because
the default values for both are defined as 0.
Example 6-2 SRST Phone Definition Commands
RemoteSite#configure terminal I
RemoteSite(config)#call-manager-fallback
RemoteSite(config-cm-fallback)#max-dn max-directory-numbers [dual-line]
[preference preference-order]
RemoteSite(config-cm-fallback)#max-ephones max-phones
128 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
The Cisco IOS command max-dn sets the maximum number of directory numbers or
virtual voice ports that can be supported by the router and activates dual-line mode. The
maximum number is platform-dependent. The default is 0.
The dual-line keyword is optional. It allows IP Phones in SRST mode to have a virtual
voice port with two channels.
NOTE The dual-line keyword facilitates call waiting, call transfer, and conference
functions by allowing two calls to occur on one line simultaneously. In dual-line mode,
all IP Phones on the Cisco Unified SRST router support two channels per virtual
voice port.
The optional parameter preference sets the global preference for creating the VoIP dial
peers for all directory numbers that are associated with the primary number. The range
is from 0 to 10. The default is 0, which is the highest preference.
NOTE The router must be rebooted to reduce the limit on the directory numbers or
virtual voice ports after the maximum allowable number is configured.
To configure the maximum number of Cisco IP Phones that an SRST router can support,
use the max-ephones command in call-manager-fallback configuration mode. The default
is 0, and the maximum configurable number is platform-dependent. The only way to
increase the maximum number of Cisco IP Phones supported is to upgrade to a higher
hardware platform.
NOTE The router must be rebooted to reduce the limit on Cisco IP Phones after the
maximum allowable number is configured.
SRST Performance Commands
To optimize performance of the system, best practice dictates that you use the limit-dn and
keepalive commands, as shown in Example 6-3.
Example 6-3 SRST Performance Commands
RemoteSite#configure terminal
RemoteSite(config)#call-manager-fallback
RemoteSite(config-cm-fallback)#limit-dn {7910 I 7935 I 7940 I 7960} max-lines
RemoteSite(config-cm-fallback)#keepalive seconds
SRST Configuration on the Cisco IOS Gateway 129
The optional Cisco IOS command limit-dn limits the directory number lines on Cisco IP
Phones during SRST mode, depending on the Cisco IP Phone model.
NOTE This command must be configured during the initial Cisco Unified SRST router
configuration, before any IP Phone actually registers with the Cisco Unified SRST router.
However, you can change the number of lines later.
The setting for the maximum number of directory lines is from 1 to 6. The default is 6. If
any active phone has the last line number greater than this limit, warning information is
displayed for phone reset.
The optional Cisco IOS command keepalive sets the time interval, in seconds, between
keepalive messages that are sent to the router by Cisco IP Phones. The range is 10 to 65535.
The default is 30.
The keepalive interval is the period of time between keepalive messages that are sent by a
network device. A keepalive message is a message that is sent by one network device to
inform another network device that the virtual circuit between the two is still active.
NOTE If the default time interval between messages of 30 seconds will be used, this
command does not have to be used.
Cisco Unified SRST Configuration Example
Figure 6-4 shows a multisite topology that supports SCCP-controlled IP Phones at the
remote SRST site. The configuration is shown in Example 6-4.
Figure 6-4 Multisite Topology Supporting SCCP-Controlled IP Phones at the Remote SRST Site
Main Site
Linel; 2001 Linel; 2002 Linel: 2003
Line2:3001 Line2:3002 Line2:3003
130 Chapter 6: Implementing Cisco Unified SRST and MGCP Fallback
NOTE More commands might be necessary, depending on the complexity of the
deployment.