Video Dial Plan Architecture
.. Corporate video conferencing was first introduced in the 1980s as a way
to help people in different cities communicate more effectively.These
first-generation solutions were based on the ITU H.320 standards
defining ISDN connection-based videoconferencing.
.. The Cisco Multimedia Conference Manager (Cisco MCM) is a
specialized Cisco IOS software image that lets network administrators
support H.323 applications on their networks without compromising
mission-critical traffic from other applications.The Cisco MCM serves
two main functions: it acts as a gatekeeper, and as a proxy.
www.syngress.com
AVVID Dial Plans • Chapter 9 331
.. A gateway is an optional element that can be implemented within the
H.323 deployment. It is an endpoint on the LAN that can provide realtime,
two-way communication between H.323 terminals or other
gateways. It is also capable of using the LAN and other ITU terminals
located on the WAN by using H.425 and Q.931 protocols.
.. A proxy gateway is a secured connection between H.323 sessions.The
Cisco Multimedia Conference Manager contains a proxy as part of its
infrastructure so it can provide QoS, traffic shaping, and security and
policy management for H.323 traffic across any secured connection.
.. The H.323 gatekeeper is an optional component capable of providing
call control services to H.323 endpoints.You may implement multiple
gatekeepers within your network, and they will remain logically separate
from the endpoints.There are currently no standards for gatekeeper-togatekeeper
communications, so you may want to explore other options
before installing multiple gatekeepers within the same segment.You
could install terminals, MCUs, gateways, or other non-H.323 LAN
devices since these may coexist in the same environment.
.. An MCU is a device that aids in getting calls to three or more endpoints
in conference type deployments. It is usually a centralized device that
assists in the facilitation of conference sessions for data, video, and/or
audio.
.. Video dial peers is a feature supported only on the MC3810 Multi-
Service Concentrator.
IT Certification CCIE,CCNP,CCIP,CCNA,CCSP,Cisco Network Optimization and Security Tips
The Role and Configuration of a Cisco CallManager and Gatekeeper
The Role and Configuration of a
Cisco CallManager and Gatekeeper
.. By implementing H.323 gatekeepers for admission control, you can
control the number of calls allowed to and from specific areas.This will
assist you in the management of bandwidth and resources for your sites
and overall infrastructure.The Cisco CallManager uses the gatekeeper to
perform admission control, especially in infrastructures that use hub and
spoke architecture for network centralization.
.. The Cisco Call Manager dial plan model requires that all Cisco
CallManagers located within a cluster be connected through an
intercluster trunk with a route pattern for each of the other clusters
within the domain.
.. The Gatekeeper dial plan model helps to clean up the overhead inherent
in the Cisco CallManager model.This is because the Cisco CallManager
only needs to maintain one intercluster trunk, known as the anonymous
device.This device is like a point-to-multipoint connection in frame
relay, as the Cisco CallManagers don’t need to be fully meshed. In this
setup, the gatekeeper is able to use the anonymous device to route calls
through the network to the correct Cisco CallManager (or cluster).
www.syngress.com
330 Chapter 9 • AVVID Dial Plans
.. The Hybrid model allows for the automatic overflow to the PSTN of
calls destined for the WAN which are unable to allocate sufficient
resources. It only needs one anonymous device for each Cisco
CallManager (cluster), thus minimizing the overhead of having to mesh
the Cisco CallManagers. It does require two routes for each destination,
however, one to the WAN and one to the PSTN.The drawback is you
need to configure the dial plan on the gatekeeper and the Cisco
CallManager.
.. For every gatekeeper located within your domain, you must configure
the intercluster CODEC you would like to use, as well as enable the
anonymous device.When that is complete, you will need to configure
the router pattern to allow calls between clusters.You would do this by
selecting a CODEC for all intercluster calls, defining the region that the
gatekeeper and cluster are located in, and select the appropriate
compression rate.
.. When configuring the Cisco CallManager gatekeeper, you are required
to enter a zone. Each Cisco CallManager will register with that zone, its
zone prefix (the directory number ranges), the bandwidth allowed for
each call admission, and the technology prefix for voice-enabled devices.
Cisco CallManager will need the gatekeeper to explicitly specify the IP
address of the Cisco CallManager within a single zone, then you must
disable the registration of all other IP address ranges so it can only exist
within that zone.
Cisco CallManager and Gatekeeper
.. By implementing H.323 gatekeepers for admission control, you can
control the number of calls allowed to and from specific areas.This will
assist you in the management of bandwidth and resources for your sites
and overall infrastructure.The Cisco CallManager uses the gatekeeper to
perform admission control, especially in infrastructures that use hub and
spoke architecture for network centralization.
.. The Cisco Call Manager dial plan model requires that all Cisco
CallManagers located within a cluster be connected through an
intercluster trunk with a route pattern for each of the other clusters
within the domain.
.. The Gatekeeper dial plan model helps to clean up the overhead inherent
in the Cisco CallManager model.This is because the Cisco CallManager
only needs to maintain one intercluster trunk, known as the anonymous
device.This device is like a point-to-multipoint connection in frame
relay, as the Cisco CallManagers don’t need to be fully meshed. In this
setup, the gatekeeper is able to use the anonymous device to route calls
through the network to the correct Cisco CallManager (or cluster).
www.syngress.com
330 Chapter 9 • AVVID Dial Plans
.. The Hybrid model allows for the automatic overflow to the PSTN of
calls destined for the WAN which are unable to allocate sufficient
resources. It only needs one anonymous device for each Cisco
CallManager (cluster), thus minimizing the overhead of having to mesh
the Cisco CallManagers. It does require two routes for each destination,
however, one to the WAN and one to the PSTN.The drawback is you
need to configure the dial plan on the gatekeeper and the Cisco
CallManager.
.. For every gatekeeper located within your domain, you must configure
the intercluster CODEC you would like to use, as well as enable the
anonymous device.When that is complete, you will need to configure
the router pattern to allow calls between clusters.You would do this by
selecting a CODEC for all intercluster calls, defining the region that the
gatekeeper and cluster are located in, and select the appropriate
compression rate.
.. When configuring the Cisco CallManager gatekeeper, you are required
to enter a zone. Each Cisco CallManager will register with that zone, its
zone prefix (the directory number ranges), the bandwidth allowed for
each call admission, and the technology prefix for voice-enabled devices.
Cisco CallManager will need the gatekeeper to explicitly specify the IP
address of the Cisco CallManager within a single zone, then you must
disable the registration of all other IP address ranges so it can only exist
within that zone.
Guidelines for the Design and Implementation of Dial Plans
Guidelines for the Design and
Implementation of Dial Plans
.. As with any project, its complexity will depend on the number of
variables factored in. Dial plan complexity can vary, based on any
number of configuration choices, such as the total amount of paths a call
can be sent through.
.. When configuring single-site campuses, you will often implement a
simple dial plan that can provide intraoffice calling (with four or five
digits depending on the site) and connections to the PSTN (usually by
dialing a 9). Long distance would also be handled by the PSTN with the
dialing party using a 9, then a 1, and the area code before dialing the
seven-digit number.
www.syngress.com
AVVID Dial Plans • Chapter 9 329
.. When you go to implement AVVID, you should work under the
assumption that the less complex it is, the better. Find out what is used
on a normal (daily) basis, and what features are seldom used.With these
answers, you can create a plan that meets the needs of the client.
.. Based on the assumption that this will be a Cisco IOS-based H.323
gateway, you would then point the router POTS dial peer to the PSTN
port (or ports) and use a destination pattern of “9” to match the leading
digit that will come from the Cisco CallManager.The match on the “9”
will make the dial peer remove the 9, so the rest of the number is
passed.
.. When creating a dial plan for a multisite WAN, you must have sufficient
resources to make it function properly. If you don’t have the proper link
bandwidth, the call will always route over the PSTN, negating the
benefits that multisite WAN connections are supposed to give you.
Implementation of Dial Plans
.. As with any project, its complexity will depend on the number of
variables factored in. Dial plan complexity can vary, based on any
number of configuration choices, such as the total amount of paths a call
can be sent through.
.. When configuring single-site campuses, you will often implement a
simple dial plan that can provide intraoffice calling (with four or five
digits depending on the site) and connections to the PSTN (usually by
dialing a 9). Long distance would also be handled by the PSTN with the
dialing party using a 9, then a 1, and the area code before dialing the
seven-digit number.
www.syngress.com
AVVID Dial Plans • Chapter 9 329
.. When you go to implement AVVID, you should work under the
assumption that the less complex it is, the better. Find out what is used
on a normal (daily) basis, and what features are seldom used.With these
answers, you can create a plan that meets the needs of the client.
.. Based on the assumption that this will be a Cisco IOS-based H.323
gateway, you would then point the router POTS dial peer to the PSTN
port (or ports) and use a destination pattern of “9” to match the leading
digit that will come from the Cisco CallManager.The match on the “9”
will make the dial peer remove the 9, so the rest of the number is
passed.
.. When creating a dial plan for a multisite WAN, you must have sufficient
resources to make it function properly. If you don’t have the proper link
bandwidth, the call will always route over the PSTN, negating the
benefits that multisite WAN connections are supposed to give you.
Creation of Calling Restrictions and Configuration of Dial Plan Groups
Creation of Calling Restrictions and
Configuration of Dial Plan Groups
.. Within Cisco CallManager, you can create calling restrictions per each
telephony device, or create closed dial plan groups (as long as they fall
within the same Cisco CallManager).What this means is that users
residing within the same Cisco CallManager can be grouped together
www.syngress.com
328 Chapter 9 • AVVID Dial Plans
with the same calling restrictions and dial plans. For example if you have
development teams that need to talk to only each other, you can restrict
their dial plans to within the group, or limit their ability to call long
distance.
.. A partition is a group of telephony devices that have similar reach ability.
These devices are composed of route patterns, IP SoftPhones, directory
numbers, and so on.
.. A calling search space is a list of partitions that can be accessed by users
in order to place a call.These calling search spaces are only allocated to
telephony devices that can start calls.With these calling search spaces
implemented, it is simple to create and use dialing restrictions, because
users are only allowed to dial those partitions in the calling search space
they are assigned to. If the user tries to call outside the allowed
partitions, they will receive a busy signal.
.. The combination of partitions and calling search spaces can allow
autonomous dial ranges on a partition basis. Extension and access codes
located within different partitions can have overlapping number
schemes, and will still work independently of each other.This is usually
seen in the implementation of a centralized call processing system. In
this example, all sites that use the same Cisco CallManager can dial the
number 9 to access the PSTN, even if they are located on different
WAN segments.
Configuration of Dial Plan Groups
.. Within Cisco CallManager, you can create calling restrictions per each
telephony device, or create closed dial plan groups (as long as they fall
within the same Cisco CallManager).What this means is that users
residing within the same Cisco CallManager can be grouped together
www.syngress.com
328 Chapter 9 • AVVID Dial Plans
with the same calling restrictions and dial plans. For example if you have
development teams that need to talk to only each other, you can restrict
their dial plans to within the group, or limit their ability to call long
distance.
.. A partition is a group of telephony devices that have similar reach ability.
These devices are composed of route patterns, IP SoftPhones, directory
numbers, and so on.
.. A calling search space is a list of partitions that can be accessed by users
in order to place a call.These calling search spaces are only allocated to
telephony devices that can start calls.With these calling search spaces
implemented, it is simple to create and use dialing restrictions, because
users are only allowed to dial those partitions in the calling search space
they are assigned to. If the user tries to call outside the allowed
partitions, they will receive a busy signal.
.. The combination of partitions and calling search spaces can allow
autonomous dial ranges on a partition basis. Extension and access codes
located within different partitions can have overlapping number
schemes, and will still work independently of each other.This is usually
seen in the implementation of a centralized call processing system. In
this example, all sites that use the same Cisco CallManager can dial the
number 9 to access the PSTN, even if they are located on different
WAN segments.
Cisco CallManager Dial Plans
Cisco CallManager Dial Plans
.. By using Cisco CallManager, you can allow for greater growth and
functionality within your network because it was designed to be
integrated with IOS gateways.
.. The creation of dial plans for internal calls to IP phones are registered
within a Cisco CallManager cluster.
.. External calls use a route pattern to direct off-network calls to a PSTN
gateway. Route patterns can also be used if there are Cisco CallManagers
located on a WAN-connected network.
.. A route pattern is the addressing method that identifies the dialed
number and uses route lists and route group configurations to determine
the route for call completion.
.. Digit manipulation involves digit removal and prefixes, digit forwarding,
and number expansion.
.. Route lists are configured to map the routes of a call to one or more
route groups.
.. Route groups allow you to control telephony devices.
.. Telephony devices are any devices capable of being connected to a route
group.
.. the digit translation table manipulates dialed digits and is supported
within Cisco Call Manager
.. Fixed-length dial peers versus Variable-length dial peers—This will help
you to decide what to use in your network.
.. Two-stage dialing occurs when a voice call is destined for the network,
and the router placing the call collects all of the dialed digits.
.. By using Cisco CallManager, you can allow for greater growth and
functionality within your network because it was designed to be
integrated with IOS gateways.
.. The creation of dial plans for internal calls to IP phones are registered
within a Cisco CallManager cluster.
.. External calls use a route pattern to direct off-network calls to a PSTN
gateway. Route patterns can also be used if there are Cisco CallManagers
located on a WAN-connected network.
.. A route pattern is the addressing method that identifies the dialed
number and uses route lists and route group configurations to determine
the route for call completion.
.. Digit manipulation involves digit removal and prefixes, digit forwarding,
and number expansion.
.. Route lists are configured to map the routes of a call to one or more
route groups.
.. Route groups allow you to control telephony devices.
.. Telephony devices are any devices capable of being connected to a route
group.
.. the digit translation table manipulates dialed digits and is supported
within Cisco Call Manager
.. Fixed-length dial peers versus Variable-length dial peers—This will help
you to decide what to use in your network.
.. Two-stage dialing occurs when a voice call is destined for the network,
and the router placing the call collects all of the dialed digits.
What Is a Dial Plan?
What Is a Dial Plan?
.. Configuring dial peers for use is essential when designing and
implementing Voice over IP on your network. Dial peers identify the
calling source and the destination points so as to define what attributes
are assigned to each call.
.. Configuring a dial peer for POTS can help you shape the deployment
of your dial peers.
.. By configuring VoIP dial peers, you can enable the router to make
outbound calls to other telephony devices located within the network.
.. Dial peers for inbound and outbound calls are used to receive and
complete calls.You must remember that the definition of inbound and
outbound is based on the perspective of the router.What this means is
that a call coming into the router is considered an inbound call and a
call originating from the router is considered an outbound call.
.. To associate a dialed string with a specific telephony device, you would
use the destination pattern.With it, the dialed string will compare itself
to the pattern and then will be routed to the voice port or the session
target (discussed later) voice network dial peer. If the call is an outbound
call, the destination pattern could also be used to filter the digits that
will be forwarded by the router to the telephony device or the PSTN.A
destination pattern must be configured for each and every POTS and
VoIP dial peer configured on the router.
.. The session target is the IP address of the router to which the call will
be directed once the dial peer is matched.
.. Route patterns (on-net) allow you to connect to multiple sites across a
WAN with connections like frame or dedicated circuits using available
network resources.
.. With Cisco CallManager, you are able to create route patterns that allow
you to route calls that differentiate between local calls and long distance
calls.
www.syngress.com
In this chapter, we talked about how a dial plan is, in its most basic form, a
system interface for telephony devices that allows users and equipment to connect
to each other by using dialing strings.These dial strings can be mapped and
routed to a multitude of locations by the controlling system.AVVID implementation
uses routers like filters to match dial peers.When a call arrives on a POTS
port, the router will match the VoIP dial peer for the outbound call. A dial plan
allows you to design your network so it can accommodate data and voice/video
within the same infrastructure. A dial plan can be created in many different ways,
depending on the needs surrounding each individual deployment.
These plans can include redundant paths, for IP networks and the PSTN, so
there will be a path to completion for calls made.
By using Cisco CallManager, you are able to allow for greater growth and
functionality within your network because Cisco CallManager was designed to
be integrated with IOS gateways to allow for greater functionality while streamlining
the installation.
We also saw that within Cisco CallManager, you can create calling restrictions
on a per telephony device, or create closed dial plan groups as long as they fall
within the same Cisco CallManager.This ensures that users residing within the
same Cisco CallManager can be grouped together with the same calling restrictions
and dial plans for ease of administration.
We also talked about the implementation of H.323 gatekeepers.These gatekeepers
handle admission control so you are able to control the number of calls
allowed to and from specific areas.We saw how this would assist you in the management
of bandwidth and resources for your sites and overall infrastructure.The
gatekeeper can also assist with address resolution.The Cisco CallManager knows
what ranges of extensions are within its control and is able to route the calls to
the proper destination. If the extension is not local to the Cisco CallManager, it
will then go to the gatekeeper for the address of another Cisco CallManager that
can direct the call to its destination.
In the LAN, it is becoming more and more feasible to introduce video and
other multiservice technologies to better leverage the local area network and its
increasing speed.With the implementation of AVVID and the usage of H.323
gateways, video to the desktop is becoming more and more commonplace.
By using Cisco AVVID video gateways, internal users can communicate more
efficiently and seamlessly.
.. Configuring dial peers for use is essential when designing and
implementing Voice over IP on your network. Dial peers identify the
calling source and the destination points so as to define what attributes
are assigned to each call.
.. Configuring a dial peer for POTS can help you shape the deployment
of your dial peers.
.. By configuring VoIP dial peers, you can enable the router to make
outbound calls to other telephony devices located within the network.
.. Dial peers for inbound and outbound calls are used to receive and
complete calls.You must remember that the definition of inbound and
outbound is based on the perspective of the router.What this means is
that a call coming into the router is considered an inbound call and a
call originating from the router is considered an outbound call.
.. To associate a dialed string with a specific telephony device, you would
use the destination pattern.With it, the dialed string will compare itself
to the pattern and then will be routed to the voice port or the session
target (discussed later) voice network dial peer. If the call is an outbound
call, the destination pattern could also be used to filter the digits that
will be forwarded by the router to the telephony device or the PSTN.A
destination pattern must be configured for each and every POTS and
VoIP dial peer configured on the router.
.. The session target is the IP address of the router to which the call will
be directed once the dial peer is matched.
.. Route patterns (on-net) allow you to connect to multiple sites across a
WAN with connections like frame or dedicated circuits using available
network resources.
.. With Cisco CallManager, you are able to create route patterns that allow
you to route calls that differentiate between local calls and long distance
calls.
www.syngress.com
In this chapter, we talked about how a dial plan is, in its most basic form, a
system interface for telephony devices that allows users and equipment to connect
to each other by using dialing strings.These dial strings can be mapped and
routed to a multitude of locations by the controlling system.AVVID implementation
uses routers like filters to match dial peers.When a call arrives on a POTS
port, the router will match the VoIP dial peer for the outbound call. A dial plan
allows you to design your network so it can accommodate data and voice/video
within the same infrastructure. A dial plan can be created in many different ways,
depending on the needs surrounding each individual deployment.
These plans can include redundant paths, for IP networks and the PSTN, so
there will be a path to completion for calls made.
By using Cisco CallManager, you are able to allow for greater growth and
functionality within your network because Cisco CallManager was designed to
be integrated with IOS gateways to allow for greater functionality while streamlining
the installation.
We also saw that within Cisco CallManager, you can create calling restrictions
on a per telephony device, or create closed dial plan groups as long as they fall
within the same Cisco CallManager.This ensures that users residing within the
same Cisco CallManager can be grouped together with the same calling restrictions
and dial plans for ease of administration.
We also talked about the implementation of H.323 gatekeepers.These gatekeepers
handle admission control so you are able to control the number of calls
allowed to and from specific areas.We saw how this would assist you in the management
of bandwidth and resources for your sites and overall infrastructure.The
gatekeeper can also assist with address resolution.The Cisco CallManager knows
what ranges of extensions are within its control and is able to route the calls to
the proper destination. If the extension is not local to the Cisco CallManager, it
will then go to the gatekeeper for the address of another Cisco CallManager that
can direct the call to its destination.
In the LAN, it is becoming more and more feasible to introduce video and
other multiservice technologies to better leverage the local area network and its
increasing speed.With the implementation of AVVID and the usage of H.323
gateways, video to the desktop is becoming more and more commonplace.
By using Cisco AVVID video gateways, internal users can communicate more
efficiently and seamlessly.
Configuring Video Dial Peers
Configuring Video Dial Peers
Video dial peers is a feature supported only on the MC3810 Multi-Service
Concentrator.What follows is a basic set of commands and how they act within
the presented command set.These are very basic, and should get you to minimal
running specifications.
1. Specify which slot the Video Dialing Module (VDM) is located in.The
keyword slot is the location value of the VDM (it will either be 1 or 2).
The keyword port is the interface.The MC3810 chassis only has one
VDM so the value is 0.
MC3810(config)# port signal slot port
2. Define the ATM dial peer for the remote system.Video dial peers will
remain until they are explicitly removed.The keyword tag identifies the
dial peer. It must be unique on the MC3810 and must be a number
from 1 to 1000.The keyword videocodec identifies the video CODEC
associated with the router.The keyword videoatm identifies the video
CODEC associated with the ATM network.
MC3810(config)# dial-peer video tag {videocodec | videoatm}
3. You must specify the E.164 address that will be associated with the dial
peer.The keywords are explained earlier in the chapter with the exception
being the + sign, which is not supported on the MC3810.
MC3810(config-dial-peer)# destination-pattern [+] string [T]
4. On the Cisco MC3810, you must configure the ATM session target for
the dial peer.The keyword serial specifies the interface for the dial peer
address.The atm keyword identifies the ATM interface number.The
interface keyword identifies the interface number.The svc nsap keyword
identifies the SVC (Switched Virtual Circuit) and the nsap (Network
Service Access Point).The nsap-address is the 40-digit hexadecimal
www.syngress.com
324 Chapter 9 • AVVID Dial Plans
number for the session target nsap.The keyword pvc is the Permanent
Virtual Circuit (PVC).The keyword name identifies the target ATM
pvc.The keyword vpi/vci is the ATM network Virtual Path Identifier
(VPI) and the Virtual Channel Identifier (VCI) for this PVC. Lastly, the
vci keyword identifies the ATM network VCI for this PVC.
MC3810(config-dial-peer0# session target {serial | atm} interface
{svc nsap nsap-address | pvc}{name | vpi/vci | vci}
From here, you would repeat these steps on the routers involved in the
configuration.
NOTE
By using PVCs to send your data traffic, you would be able to identify a
PVC defined on the ATM interface as a session target by using a name or
VPI/VCI pair.
www.syngress.
Video dial peers is a feature supported only on the MC3810 Multi-Service
Concentrator.What follows is a basic set of commands and how they act within
the presented command set.These are very basic, and should get you to minimal
running specifications.
1. Specify which slot the Video Dialing Module (VDM) is located in.The
keyword slot is the location value of the VDM (it will either be 1 or 2).
The keyword port is the interface.The MC3810 chassis only has one
VDM so the value is 0.
MC3810(config)# port signal slot port
2. Define the ATM dial peer for the remote system.Video dial peers will
remain until they are explicitly removed.The keyword tag identifies the
dial peer. It must be unique on the MC3810 and must be a number
from 1 to 1000.The keyword videocodec identifies the video CODEC
associated with the router.The keyword videoatm identifies the video
CODEC associated with the ATM network.
MC3810(config)# dial-peer video tag {videocodec | videoatm}
3. You must specify the E.164 address that will be associated with the dial
peer.The keywords are explained earlier in the chapter with the exception
being the + sign, which is not supported on the MC3810.
MC3810(config-dial-peer)# destination-pattern [+] string [T]
4. On the Cisco MC3810, you must configure the ATM session target for
the dial peer.The keyword serial specifies the interface for the dial peer
address.The atm keyword identifies the ATM interface number.The
interface keyword identifies the interface number.The svc nsap keyword
identifies the SVC (Switched Virtual Circuit) and the nsap (Network
Service Access Point).The nsap-address is the 40-digit hexadecimal
www.syngress.com
324 Chapter 9 • AVVID Dial Plans
number for the session target nsap.The keyword pvc is the Permanent
Virtual Circuit (PVC).The keyword name identifies the target ATM
pvc.The keyword vpi/vci is the ATM network Virtual Path Identifier
(VPI) and the Virtual Channel Identifier (VCI) for this PVC. Lastly, the
vci keyword identifies the ATM network VCI for this PVC.
MC3810(config-dial-peer0# session target {serial | atm} interface
{svc nsap nsap-address | pvc}{name | vpi/vci | vci}
From here, you would repeat these steps on the routers involved in the
configuration.
NOTE
By using PVCs to send your data traffic, you would be able to identify a
PVC defined on the ATM interface as a session target by using a name or
VPI/VCI pair.
www.syngress.
Multipoint Control Units
Multipoint Control Units
So, what is a Multipoint Control Unit (MCU)? An MCU is a device that aids in
getting calls out to three or more endpoints in conference type deployments. It
works like a meet-me call bridge, and is usually a centralized device that assists in
the facilitation of conference sessions for data, video, and/or audio.
So, what is a Multipoint Control Unit (MCU)? An MCU is a device that aids in
getting calls out to three or more endpoints in conference type deployments. It
works like a meet-me call bridge, and is usually a centralized device that assists in
the facilitation of conference sessions for data, video, and/or audio.
Bandwidth control
Bandwidth control The gatekeeper supports BRQ/BRJ/BCF messages
based upon bandwidth management.This can also be set as a null
function that accepts all requests for bandwidth changes.
based upon bandwidth management.This can also be set as a null
function that accepts all requests for bandwidth changes.
Zone management
Zone management Used for registered terminals, Multipoint Control
Units, and gateways.
Units, and gateways.
Admission control
Admission control This authorizes access using ARQ/ACF/ARJ/
H225.0 messages, based upon call authorization, bandwidth, or other
criteria set by the vendor and your predefined settings.You can also set
this as a null function to admit all requests without performing filtering.
H225.0 messages, based upon call authorization, bandwidth, or other
criteria set by the vendor and your predefined settings.You can also set
this as a null function to admit all requests without performing filtering.
Address translation
Address translation Alias addresses can be maintained to transport
address translation.The gatekeeper often uses a translation table that is
updated by using registration messages (other table update methods are
allowed as well, such as manual updates).
address translation.The gatekeeper often uses a translation table that is
updated by using registration messages (other table update methods are
allowed as well, such as manual updates).
The H.323 Gatekeeper
The H.323 Gatekeeper
The H.323 Gatekeeper is an optional component that provides call control services
for H.323 endpoints within your network.You can implement multiple
gatekeepers to run within your network, so long as they remain logically separate
from the endpoints.There are currently no standards for gatekeeper-to-gatekeeper
communications as of yet, so you may want to explore other options
before installing multiple gatekeepers within the same segment. Keep in mind,
this can cause some implementation issues, and, if implemented within your main
LAN/WAN, some network issues as well.To alleviate these problems, you could
install terminals, Multipoint Control Units, gateways, or other non-H.323 LAN
devices since these can coexist in the same environment. H.323 Gatekeepers provide
the following functions:
The H.323 Gatekeeper is an optional component that provides call control services
for H.323 endpoints within your network.You can implement multiple
gatekeepers to run within your network, so long as they remain logically separate
from the endpoints.There are currently no standards for gatekeeper-to-gatekeeper
communications as of yet, so you may want to explore other options
before installing multiple gatekeepers within the same segment. Keep in mind,
this can cause some implementation issues, and, if implemented within your main
LAN/WAN, some network issues as well.To alleviate these problems, you could
install terminals, Multipoint Control Units, gateways, or other non-H.323 LAN
devices since these can coexist in the same environment. H.323 Gatekeepers provide
the following functions:
Proxy Gateway
Proxy Gateway
A proxy gateway is a secured connection between H.323 sessions, allowing Cisco
Multimedia Conference Manager to contain a proxy as part of its infrastructure
so it can provide QoS, traffic shaping, and security and policy management for
H.323 traffic across any secured connection. A proxy in this case is very similar to
www.syngress.com
322 Chapter 9 • AVVID Dial Plans
proxy gateways in other network deployments. In essence, it is a go-between that
is able to connect two dissimilar connections, or two connections that need to be
separated. It can establish and maintain its connection between these multiple end
points, adding to your overall efficiency by offloading some of the features to the
proxy gateway.
A proxy gateway is a secured connection between H.323 sessions, allowing Cisco
Multimedia Conference Manager to contain a proxy as part of its infrastructure
so it can provide QoS, traffic shaping, and security and policy management for
H.323 traffic across any secured connection. A proxy in this case is very similar to
www.syngress.com
322 Chapter 9 • AVVID Dial Plans
proxy gateways in other network deployments. In essence, it is a go-between that
is able to connect two dissimilar connections, or two connections that need to be
separated. It can establish and maintain its connection between these multiple end
points, adding to your overall efficiency by offloading some of the features to the
proxy gateway.
Gateway
Gateway
A gateway is an optional element that can be implemented within the H.323
deployment. It is an endpoint on the LAN that can provide real-time, two-way
communication between H.323 terminals or other gateways. It is also capable of
using the LAN and other ITU terminals located on the WAN by using H.425
and Q.931 protocols.These gateways are not required if there are no connections
to other networks.You will use these gateways when you need to:
Create connections over analog PSTN terminals
Create connections with H.320-based terminals across ISDN circuits
Create connections with H.324-based terminals over a PSTN networks
Gateways can translate between H.323 conferencing endpoints and compliant
terminals, H.225 to H.221, or between communication procedures, H.245 to
H.242.These gateways can also translate between audio and video CODECs,
performing call setup and termination on both sides of a network. If you configure
the gateway properly, it can also support H.310, H. 321, H.322, and V.70
standard terminals.
A gateway is an optional element that can be implemented within the H.323
deployment. It is an endpoint on the LAN that can provide real-time, two-way
communication between H.323 terminals or other gateways. It is also capable of
using the LAN and other ITU terminals located on the WAN by using H.425
and Q.931 protocols.These gateways are not required if there are no connections
to other networks.You will use these gateways when you need to:
Create connections over analog PSTN terminals
Create connections with H.320-based terminals across ISDN circuits
Create connections with H.324-based terminals over a PSTN networks
Gateways can translate between H.323 conferencing endpoints and compliant
terminals, H.225 to H.221, or between communication procedures, H.245 to
H.242.These gateways can also translate between audio and video CODECs,
performing call setup and termination on both sides of a network. If you configure
the gateway properly, it can also support H.310, H. 321, H.322, and V.70
standard terminals.
The Cisco Multimedia Conference Manager
The Cisco Multimedia Conference Manager
The Cisco Multimedia Conference Manager (Cisco MCM) is a specialized Cisco
IOS software image that lets network administrators support H.323 applications
on their networks without compromising mission-critical traffic from other
applications.The Cisco MCM serves two main functions: it acts as a gatekeeper,
and as a proxy.The following sections will help you understand what the differences
are between these two functions.
The Cisco Multimedia Conference Manager (Cisco MCM) is a specialized Cisco
IOS software image that lets network administrators support H.323 applications
on their networks without compromising mission-critical traffic from other
applications.The Cisco MCM serves two main functions: it acts as a gatekeeper,
and as a proxy.The following sections will help you understand what the differences
are between these two functions.
Video Dial Plan Architecture
Video Dial Plan Architecture
Technically, there is a lot of overlap between the configuration of voice and video
on the network.The following sections are written to explain some of the differences,
but keep in mind that topics discussed earlier within the chapter still need
to be considered when implementing a video dial plan.
When corporate videoconferencing was first introduced in the 1980s, people
saw it as a way to help connect people located in different cities so they could
communicate more effectively and efficiently, and at a greater savings in cost.
www.syngress.com
Figure 9.7 Configuration of a Group with Cisco CallManager
320 Chapter 9 • AVVID Dial Plans
With this in mind, there came the creation of standards that would allow these
remote locations to connect to each other.These first-generation solutions were
based upon the International Telecommunications Union (ITU) H.320 standards
defining Integrated Switched Digital Network (ISDN) connection-based videoconferencing.
Not long after this, second-generation solutions were created, bringing videoconferencing
to computer desktops.The main drawback to this more refined
technology was that it was still dependent upon ISDN and expensive CODEC
devices, so it wasn’t feasible to offer it to most users for normal business needs. In
the mid-1990s, the creation of the third-generation, LAN-based solutions,
became more prevalent in many organizations. Accessible desktop videoconferencing
applications multiplied, but until recently, these had remained proprietary,
and quite often, very expensive.
Because of compatibility issues between different device and software solutions,
widespread deployment up to this time has been severely limited.With
recent changes in world travel, however, you may begin to see videoconferencing
become more prevalent in the next few years; a trend that should, in turn, lead to
increased attempts at standardization as more companies try to corner the market.
You have to remember that when H.323 and H.324 standards were created,
they gave software and hardware vendors the ability to create videoconferencing
packages that were more manageable and affordable.This isn’t to say they solved
all problems are associated with videoconferencing (circuits are not always cheap,
and do not necessarily offer the uptime and dedicated bandwidth necessary to
support every organization).
The H.323 standard was able to define videoconferencing technologies and
enabled multivendor interoperability for the first time. Meanwhile, the H.324
standard was defined as a solution for videoconferencing using POTS lines.When
Cisco Systems started to deploy AVVID, they were able to enable end-to-end,
global desktop videoconferencing,VoIP, multilocation collaboration, and electronic
whiteboard applications.
As more and more applications based upon H.323 become available, and the
cost of digital video cameras continues to decline, there will be a more ubiquitous
implementation of desktop video conferencing. Also, as companies become
more and more economically savvy, you will see many accountants wanting to
leverage existing technologies and add one time costs (such as the digital camera)
to help with the bottom line and maintain cash flow.
Cisco introduced its Multimedia Conference Manager in February of 2001.
It is an H.323-compliant software that enables its users to create policy-based
www.syngress.com
AVVID Dial Plans • Chapter 9 321
management for H.323 applications, and is one of the first multifunctional, scalable
solutions available for a large market. It allows you to control and manage
voice and video connections with much more efficiency given your network
resources.
Technically, there is a lot of overlap between the configuration of voice and video
on the network.The following sections are written to explain some of the differences,
but keep in mind that topics discussed earlier within the chapter still need
to be considered when implementing a video dial plan.
When corporate videoconferencing was first introduced in the 1980s, people
saw it as a way to help connect people located in different cities so they could
communicate more effectively and efficiently, and at a greater savings in cost.
www.syngress.com
Figure 9.7 Configuration of a Group with Cisco CallManager
320 Chapter 9 • AVVID Dial Plans
With this in mind, there came the creation of standards that would allow these
remote locations to connect to each other.These first-generation solutions were
based upon the International Telecommunications Union (ITU) H.320 standards
defining Integrated Switched Digital Network (ISDN) connection-based videoconferencing.
Not long after this, second-generation solutions were created, bringing videoconferencing
to computer desktops.The main drawback to this more refined
technology was that it was still dependent upon ISDN and expensive CODEC
devices, so it wasn’t feasible to offer it to most users for normal business needs. In
the mid-1990s, the creation of the third-generation, LAN-based solutions,
became more prevalent in many organizations. Accessible desktop videoconferencing
applications multiplied, but until recently, these had remained proprietary,
and quite often, very expensive.
Because of compatibility issues between different device and software solutions,
widespread deployment up to this time has been severely limited.With
recent changes in world travel, however, you may begin to see videoconferencing
become more prevalent in the next few years; a trend that should, in turn, lead to
increased attempts at standardization as more companies try to corner the market.
You have to remember that when H.323 and H.324 standards were created,
they gave software and hardware vendors the ability to create videoconferencing
packages that were more manageable and affordable.This isn’t to say they solved
all problems are associated with videoconferencing (circuits are not always cheap,
and do not necessarily offer the uptime and dedicated bandwidth necessary to
support every organization).
The H.323 standard was able to define videoconferencing technologies and
enabled multivendor interoperability for the first time. Meanwhile, the H.324
standard was defined as a solution for videoconferencing using POTS lines.When
Cisco Systems started to deploy AVVID, they were able to enable end-to-end,
global desktop videoconferencing,VoIP, multilocation collaboration, and electronic
whiteboard applications.
As more and more applications based upon H.323 become available, and the
cost of digital video cameras continues to decline, there will be a more ubiquitous
implementation of desktop video conferencing. Also, as companies become
more and more economically savvy, you will see many accountants wanting to
leverage existing technologies and add one time costs (such as the digital camera)
to help with the bottom line and maintain cash flow.
Cisco introduced its Multimedia Conference Manager in February of 2001.
It is an H.323-compliant software that enables its users to create policy-based
www.syngress.com
AVVID Dial Plans • Chapter 9 321
management for H.323 applications, and is one of the first multifunctional, scalable
solutions available for a large market. It allows you to control and manage
voice and video connections with much more efficiency given your network
resources.
Gatekeeper Configuration
Configuring a Cisco CallManager Device Pool
Figure 9.6 Configuration of a Region with Cisco CallManager
Configuration of a Group with Cisco CallManager
Gatekeeper Configuration
When configuring the Cisco CallManager gatekeeper, you are required to enter a
zone. Each Cisco CallManager will register with that zone: its zone prefix (this is
the directory number ranges), the bandwidth allowed for each call admission, and
the technology prefix for voice-enabled devices. Cisco CallManager will need the
gatekeeper to explicitly specify the IP address of the Cisco CallManager within a
single zone, and then you must disable the registration of all other IP address
ranges so it only exists within that zone. Cisco CallManager will register with the
gatekeeper using its IP address as its H.323 ID.
Figure 9.6 Configuration of a Region with Cisco CallManager
Configuration of a Group with Cisco CallManager
Gatekeeper Configuration
When configuring the Cisco CallManager gatekeeper, you are required to enter a
zone. Each Cisco CallManager will register with that zone: its zone prefix (this is
the directory number ranges), the bandwidth allowed for each call admission, and
the technology prefix for voice-enabled devices. Cisco CallManager will need the
gatekeeper to explicitly specify the IP address of the Cisco CallManager within a
single zone, and then you must disable the registration of all other IP address
ranges so it only exists within that zone. Cisco CallManager will register with the
gatekeeper using its IP address as its H.323 ID.
The Hybrid Model
The Hybrid Model
This type of configuration incorporates the best of both worlds.There is automatic
overflow to the PSTN for calls destined for the WAN which are unable to
allocate sufficient resources. It only needs one anonymous device for each Cisco
CallManager (cluster), thus minimizing the overhead of having to mesh the Cisco
CallManagers. It does require two routes for each destination, however, one to
the WAN and one to the PSTN.The drawback is you need to configure the dial
plan on the gatekeeper and the Cisco CallManager.
This type of configuration incorporates the best of both worlds.There is automatic
overflow to the PSTN for calls destined for the WAN which are unable to
allocate sufficient resources. It only needs one anonymous device for each Cisco
CallManager (cluster), thus minimizing the overhead of having to mesh the Cisco
CallManagers. It does require two routes for each destination, however, one to
the WAN and one to the PSTN.The drawback is you need to configure the dial
plan on the gatekeeper and the Cisco CallManager.
The Gatekeeper Model
The Gatekeeper Model
This dial plan model helps clean up the overhead inherent within the Cisco
CallManager model.This is because the Cisco CallManager only needs to maintain
one intercluster trunk known as the anonymous device.This device is like a
point-to-multipoint connection in frame relay, as the Cisco CallManagers don’t
need to be fully meshed. In this set up, the gatekeeper is able to use the anonymous
device to route calls through the network to the correct Cisco
CallManager (or cluster).
www.syngress.com
AVVID Dial Plans • Chapter 9 317
You can set this model up for exclusionary clusters.What this means is that
there is no automatic overflow for calls destined for the WAN which are unable
to complete due to insufficient resources. In this configuration, you need two
route patterns, one to the WAN and one to the PSTN.When users are unable to
connect over the WAN, they need to dial the PSTN number to access the other
site.The dial plans are configured and executed within the gatekeeper, so when
you add new Cisco CallManagers, or change the dial plan, you only need to configure
the gatekeeper.
This dial plan model helps clean up the overhead inherent within the Cisco
CallManager model.This is because the Cisco CallManager only needs to maintain
one intercluster trunk known as the anonymous device.This device is like a
point-to-multipoint connection in frame relay, as the Cisco CallManagers don’t
need to be fully meshed. In this set up, the gatekeeper is able to use the anonymous
device to route calls through the network to the correct Cisco
CallManager (or cluster).
www.syngress.com
AVVID Dial Plans • Chapter 9 317
You can set this model up for exclusionary clusters.What this means is that
there is no automatic overflow for calls destined for the WAN which are unable
to complete due to insufficient resources. In this configuration, you need two
route patterns, one to the WAN and one to the PSTN.When users are unable to
connect over the WAN, they need to dial the PSTN number to access the other
site.The dial plans are configured and executed within the gatekeeper, so when
you add new Cisco CallManagers, or change the dial plan, you only need to configure
the gatekeeper.
The Cisco CallManager Model
The Cisco CallManager Model
This dial plan model requires that all Cisco CallManagers located within a cluster
be connected through an intercluster trunk with a route pattern for each of the
other clusters within the domain.This creates quite a bit of overhead as you add
new Cisco CallManager clusters since you will have to reconfigure the Cisco
CallManager every time a new cluster is added or a dial plan changes.With this
deployment, the gatekeeper handles call admission control but not dial plan resolution.
This setup is pretty close to what you might think of as a normal PBX.
The Cisco CallManager will need two routes configured for each destination,
one for the WAN and one for the PSTN. Calls are then automatically routed
over the PSTN if there are insufficient resources on the WAN.
This dial plan model requires that all Cisco CallManagers located within a cluster
be connected through an intercluster trunk with a route pattern for each of the
other clusters within the domain.This creates quite a bit of overhead as you add
new Cisco CallManager clusters since you will have to reconfigure the Cisco
CallManager every time a new cluster is added or a dial plan changes.With this
deployment, the gatekeeper handles call admission control but not dial plan resolution.
This setup is pretty close to what you might think of as a normal PBX.
The Cisco CallManager will need two routes configured for each destination,
one for the WAN and one for the PSTN. Calls are then automatically routed
over the PSTN if there are insufficient resources on the WAN.
The Role and Configuration of a Cisco CallManager and Gatekeeper
The Role and Configuration of a Cisco
CallManager and Gatekeeper
By implementing H.323 gatekeepers for admission control, you will be able to
control the number of calls allowed to and from specific areas.This will assist you
in the management of bandwidth and resources for your sites and overall infrastructure.
The Cisco CallManager uses the gatekeeper to perform admission control,
especially in infrastructures that use hub and spoke architecture for network
centralization.
www.syngress.com
316 Chapter 9 • AVVID Dial Plans
The gatekeeper will also assist with address resolution.The Cisco
CallManager will know what ranges of extensions are within its control and is
able to route the calls to the proper destination. If the extension is not local to
the Cisco CallManager, it goes to the gatekeeper for the address of another Cisco
CallManager that can direct the call to its destination (hopefully).
The reason this occurs is because each Cisco CallManager registers itself with
the gatekeeper, which is configured with the address range of each Cisco
CallManager (or cluster).This helps hold the dial plans each Cisco CallManager
will need to set up.There are three types of dial plan deployment models for destination
resolution of calls from within the cluster.They are:
The Cisco CallManager Model
The Gatekeeper Model
The Hybrid Model
CallManager and Gatekeeper
By implementing H.323 gatekeepers for admission control, you will be able to
control the number of calls allowed to and from specific areas.This will assist you
in the management of bandwidth and resources for your sites and overall infrastructure.
The Cisco CallManager uses the gatekeeper to perform admission control,
especially in infrastructures that use hub and spoke architecture for network
centralization.
www.syngress.com
316 Chapter 9 • AVVID Dial Plans
The gatekeeper will also assist with address resolution.The Cisco
CallManager will know what ranges of extensions are within its control and is
able to route the calls to the proper destination. If the extension is not local to
the Cisco CallManager, it goes to the gatekeeper for the address of another Cisco
CallManager that can direct the call to its destination (hopefully).
The reason this occurs is because each Cisco CallManager registers itself with
the gatekeeper, which is configured with the address range of each Cisco
CallManager (or cluster).This helps hold the dial plans each Cisco CallManager
will need to set up.There are three types of dial plan deployment models for destination
resolution of calls from within the cluster.They are:
The Cisco CallManager Model
The Gatekeeper Model
The Hybrid Model
Configuration of a Cisco CallManager Cluster
Configuration of a Cisco CallManager Cluster
For every gatekeeper located within your domain, you must configure an intercluster
CODEC you would like to use, and along with this enable the anonymous
device.When that is complete, you will need to configure the router
pattern to allow calls between clusters.You do this by selecting a CODEC for all
intercluster calls, defining the region the gatekeeper and cluster are located in,
and selecting the appropriate compression rate.
When the Cisco CallManager registers with the gatekeeper, it identifies itself
as a VoIP gateway that has a technology prefix of Voice.You can manually set this
configuration by going to the Service tab, scrolling down to the Service
Parameters, and from there updating the Cisco CallManager service parameter
GateKeeperSupportedPrefix configuration. Figures 9.5, 9.6, and 9.7 illustrate the
basic configuration of a Cisco CallManger.
NOTE
The GateKeeperSupportedPrefix is hidden by default. Refer to your Cisco
documentation regarding how to access it.
For every gatekeeper located within your domain, you must configure an intercluster
CODEC you would like to use, and along with this enable the anonymous
device.When that is complete, you will need to configure the router
pattern to allow calls between clusters.You do this by selecting a CODEC for all
intercluster calls, defining the region the gatekeeper and cluster are located in,
and selecting the appropriate compression rate.
When the Cisco CallManager registers with the gatekeeper, it identifies itself
as a VoIP gateway that has a technology prefix of Voice.You can manually set this
configuration by going to the Service tab, scrolling down to the Service
Parameters, and from there updating the Cisco CallManager service parameter
GateKeeperSupportedPrefix configuration. Figures 9.5, 9.6, and 9.7 illustrate the
basic configuration of a Cisco CallManger.
NOTE
The GateKeeperSupportedPrefix is hidden by default. Refer to your Cisco
documentation regarding how to access it.
Creating a Dial Plan for a Multisite Organization
Creating a Dial Plan for a
Multisite Organization
For the creation of a Dial Plan for a multisite WAN, you must have sufficient
resources to make it function properly. If you don’t have the proper link bandwidth,
the call will always route over the PSTN, therefore negating the benefits
that multisite WAN connections are supposed to give you. If the system can route
calls over the WAN then you will be able to bundle the costs of long distance
with your data connection, and hopefully maximize your investment.When you
implement the dial plan with the proper gatekeeper call admission control mechanism,
the dial plan will decide when to place calls across the WAN and what it
will do if the gatekeeper will not accept the call.
Multisite Organization
For the creation of a Dial Plan for a multisite WAN, you must have sufficient
resources to make it function properly. If you don’t have the proper link bandwidth,
the call will always route over the PSTN, therefore negating the benefits
that multisite WAN connections are supposed to give you. If the system can route
calls over the WAN then you will be able to bundle the costs of long distance
with your data connection, and hopefully maximize your investment.When you
implement the dial plan with the proper gatekeeper call admission control mechanism,
the dial plan will decide when to place calls across the WAN and what it
will do if the gatekeeper will not accept the call.
Verifying the Dial Plan is Correct
Verifying the Dial Plan is Correct
Based on the assumption that this will be a Cisco IOS-based H.323 gateway, you
would then point the router POTS dial peer to the PSTN port (or ports) and
use a destination-pattern of “9” to match the leading digit that will come from
the Cisco CallManager.The match on the “9” will make the dial peer remove the
9, so the rest of the number is passed.
dial-peer voice 100 pots
direct-inward-dial \\Creates the DID for incoming calls
!
destination-pattern 9 \\Removes the 9 when a call is placed
!
port 1/0:15 \\This will direct the call to the PRI port
\\ 1/0
!
!
www.syngress.com
AVVID Dial Plans • Chapter 9 315
You will not need multiple POTS dial peers if you are not setting up a hunt
group that will use multiple POTS. If you were to set up multiple POTS for
hunt groups, and you wanted the calls to cycle through several configured voice
ports, it might look like this:
!
dial-peer voice 100 pots
destination-pattern 9
port 1/0/0
!
dial-peer voice 101 pots
destination-pattern 9
port 1/0/1
!
Based on the assumption that this will be a Cisco IOS-based H.323 gateway, you
would then point the router POTS dial peer to the PSTN port (or ports) and
use a destination-pattern of “9” to match the leading digit that will come from
the Cisco CallManager.The match on the “9” will make the dial peer remove the
9, so the rest of the number is passed.
dial-peer voice 100 pots
direct-inward-dial \\Creates the DID for incoming calls
!
destination-pattern 9 \\Removes the 9 when a call is placed
!
port 1/0:15 \\This will direct the call to the PRI port
\\ 1/0
!
!
www.syngress.com
AVVID Dial Plans • Chapter 9 315
You will not need multiple POTS dial peers if you are not setting up a hunt
group that will use multiple POTS. If you were to set up multiple POTS for
hunt groups, and you wanted the calls to cycle through several configured voice
ports, it might look like this:
!
dial-peer voice 100 pots
destination-pattern 9
port 1/0/0
!
dial-peer voice 101 pots
destination-pattern 9
port 1/0/1
!
Configuring the Dial Plan within the Cisco CallManager
Configuring the Dial Plan within the Cisco CallManager
Configuring the dial plan within CallManager is a fairly straightforward process,
which can be distilled down to these easy steps:
1. Enter an access code of 9. as the access code delimiter.You can add the
route pattern digits, wildcard matches, or the “@” character for North
America.
2. Ensure that the Route this pattern and Provide secondary dial
tone options are set.
3. Point the route pattern to a gateway device (H.323, MGCP, Skinny
Gateway Protocol, SIP).
4. If the gateway device is MGCP or Skinny Gateway Protocol, you must
make sure the access code is discarded.To do this, set discard digits to
under Called Party Transformations.
5. If the gateway device is a Cisco IOS-based H.323 gateway, then the
access code will need to be passed along with the called digits.To do
this, set discard digits to under Called Party Transformations.
6. Create the route pattern in the database.
Configuring the dial plan within CallManager is a fairly straightforward process,
which can be distilled down to these easy steps:
1. Enter an access code of 9. as the access code delimiter.You can add the
route pattern digits, wildcard matches, or the “@” character for North
America.
2. Ensure that the Route this pattern and Provide secondary dial
tone options are set.
3. Point the route pattern to a gateway device (H.323, MGCP, Skinny
Gateway Protocol, SIP).
4. If the gateway device is MGCP or Skinny Gateway Protocol, you must
make sure the access code is discarded.To do this, set discard digits to
5. If the gateway device is a Cisco IOS-based H.323 gateway, then the
access code will need to be passed along with the called digits.To do
this, set discard digits to
6. Create the route pattern in the database.
Sample Dial Plan Route Pattern Comments
Sample Dial Plan
Route Pattern Comments
9.911 No Pattern Standard number for emergency calls
9.411 No Pattern Standard number for Information
0.5551212 No Pattern Standard number for Information
9.1[800]XXXXXXX 1800XXXXXXX Toll-Free Call
9.1[866]XXXXXXX 1866XXXXXXX Toll-Free Call
9.1[877]XXXXXXX 1877XXXXXXX Toll-Free Call
9.555XXXX No Pattern Local Exchange Numbers—seven-digit
numbers
00! No Pattern International—uses interdigit default
timeout (ten seconds)
00!# No Pattern International—uses # to signal end of
dial character
Route Pattern Comments
9.911 No Pattern Standard number for emergency calls
9.411 No Pattern Standard number for Information
0.5551212 No Pattern Standard number for Information
9.1[800]XXXXXXX 1800XXXXXXX Toll-Free Call
9.1[866]XXXXXXX 1866XXXXXXX Toll-Free Call
9.1[877]XXXXXXX 1877XXXXXXX Toll-Free Call
9.555XXXX No Pattern Local Exchange Numbers—seven-digit
numbers
00! No Pattern International—uses interdigit default
timeout (ten seconds)
00!# No Pattern International—uses # to signal end of
dial character
Design Considerations for the Creation of a Dial Plan
Design Considerations for the Creation of a Dial Plan
For this example, we will discuss the implementation of a national dial plan for a
location that resides within the United States.This methodology can be used to
create other plans anywhere in the world if you implement the proper dialing
schema.
For this example, we will discuss the implementation of a national dial plan for a
location that resides within the United States.This methodology can be used to
create other plans anywhere in the world if you implement the proper dialing
schema.
Cisco CallManager Flow Chart for Single Campuses Route Pattern
Cisco CallManager Flow Chart for Single Campuses
Route Pattern
9.@
Route List
Local PSTN
Route Group
Gateway 2
Route Group
Gateway 1
Calling
Party
(A)
Destination
(B)
PSTN
Digit Manipulation PSTN
(Removal of Dialed Access Code)
Digit Manipulation
(Removal of Dialed Access Code)
This Gateway is configured
as the Primary Gateway.
This Gateway is configured
as the Secondary Gateway.
The dial plan is set up to use one route pattern.The 9.@ is the configuration
pattern that signifies the 9 as the access code to connect to the PSTN.The @ is
required to configure the dialing plan as the North American standard (E.164).
The “.” is used by the Cisco CallManager to tell it which digits are considered
after the access code.This needs to be configured to be sure to remove the correct
digits (the digits located on the left of the dot).
The route pattern will also allow the dialing of 911 for emergency services.
The route group is configured to remove the access code (9) from the dialed
string so the call can be properly routed through the PSTN.
You would often see the multiple PSTN setup for redundancy.This way, if
one PSTN becomes unavailable, or the gateway connected to the PSTN does not
function, the Cisco CallManager will route the call through the secondary
gateway.
www.syngress.com
Figure 9.4 Cisco CallManager Flow Chart for Single Campuses
Route Pattern
9.@
Route List
Local PSTN
Route Group
Gateway 2
Route Group
Gateway 1
Calling
Party
(A)
Destination
(B)
PSTN
Digit Manipulation PSTN
(Removal of Dialed Access Code)
Digit Manipulation
(Removal of Dialed Access Code)
This Gateway is configured
as the Primary Gateway.
This Gateway is configured
as the Secondary Gateway.
AVVID Dial Plans • Chapter 9 311
When configuring a Cisco IOS H.323 gateway, try to minimize the number
of entries. For the most part, the dial plan configurations should occur at the
Cisco CallManager.This adds to the efficiency of the router.You could also configure
these gateways to use the Skinny Gateway Protocol or MGCP, but you will
more commonly use the H.323-based gateways.
dial-peer voice 1 voip
codec g711ulaw \\This states that the Dial peer for
\\ all incoming calls from PSTN to
\\ Cisco CallManager's IP address must
\\be G.711
dtmf-relay h245-alphanumeric
destination-pattern 9....
session target ipv4:10.1.100.1 \\This is the Cisco CallManager's IP
\\address
!
dial-peer voice 2 pots \\This is the Dial peer for all 7-digit
\\outgoing PSTN numbers
destination-pattern......
port 1/0:1
!
dial-peer voice 3 pots \\This is the Dial peer for all 10-
\\digit outgoing PSTN numbers
destination-pattern 1.......
prefix 1
port 1/0:1
!
dial-peer voice 4 pots \\This is the Dial peer for 911
\\services
destination-pattern 911
prefix 911
port 1/0:1
With this configuration, the Cisco CallManager assumes the 1 + 10 digit dial
string is required to make long distance calls through the PSTN, and that sevendigit
calling would use the PSTN for local calls. Even though the addition of the
www.syngress.com
312 Chapter 9 • AVVID Dial Plans
9.@ (as discussed earlier) includes the ability to dial 911 for emergencies, the Cisco
IOS gateway requires an entry for the dial peer.You could also add dial peers for
411 if you want that service to be available (this is also handled by the 9.@).
NOTE
Voice over IP points to layer three (routed) addresses, so it looks like IP
packet-based traffic as it traverses the network. POTS traffic (and VoFR)
is based on layer two, so it is treated as a voice call all the way to
completion.
Route Pattern
9.@
Route List
Local PSTN
Route Group
Gateway 2
Route Group
Gateway 1
Calling
Party
(A)
Destination
(B)
PSTN
Digit Manipulation PSTN
(Removal of Dialed Access Code)
Digit Manipulation
(Removal of Dialed Access Code)
This Gateway is configured
as the Primary Gateway.
This Gateway is configured
as the Secondary Gateway.
The dial plan is set up to use one route pattern.The 9.@ is the configuration
pattern that signifies the 9 as the access code to connect to the PSTN.The @ is
required to configure the dialing plan as the North American standard (E.164).
The “.” is used by the Cisco CallManager to tell it which digits are considered
after the access code.This needs to be configured to be sure to remove the correct
digits (the digits located on the left of the dot).
The route pattern will also allow the dialing of 911 for emergency services.
The route group is configured to remove the access code (9) from the dialed
string so the call can be properly routed through the PSTN.
You would often see the multiple PSTN setup for redundancy.This way, if
one PSTN becomes unavailable, or the gateway connected to the PSTN does not
function, the Cisco CallManager will route the call through the secondary
gateway.
www.syngress.com
Figure 9.4 Cisco CallManager Flow Chart for Single Campuses
Route Pattern
9.@
Route List
Local PSTN
Route Group
Gateway 2
Route Group
Gateway 1
Calling
Party
(A)
Destination
(B)
PSTN
Digit Manipulation PSTN
(Removal of Dialed Access Code)
Digit Manipulation
(Removal of Dialed Access Code)
This Gateway is configured
as the Primary Gateway.
This Gateway is configured
as the Secondary Gateway.
AVVID Dial Plans • Chapter 9 311
When configuring a Cisco IOS H.323 gateway, try to minimize the number
of entries. For the most part, the dial plan configurations should occur at the
Cisco CallManager.This adds to the efficiency of the router.You could also configure
these gateways to use the Skinny Gateway Protocol or MGCP, but you will
more commonly use the H.323-based gateways.
dial-peer voice 1 voip
codec g711ulaw \\This states that the Dial peer for
\\ all incoming calls from PSTN to
\\ Cisco CallManager's IP address must
\\be G.711
dtmf-relay h245-alphanumeric
destination-pattern 9....
session target ipv4:10.1.100.1 \\This is the Cisco CallManager's IP
\\address
!
dial-peer voice 2 pots \\This is the Dial peer for all 7-digit
\\outgoing PSTN numbers
destination-pattern......
port 1/0:1
!
dial-peer voice 3 pots \\This is the Dial peer for all 10-
\\digit outgoing PSTN numbers
destination-pattern 1.......
prefix 1
port 1/0:1
!
dial-peer voice 4 pots \\This is the Dial peer for 911
\\services
destination-pattern 911
prefix 911
port 1/0:1
With this configuration, the Cisco CallManager assumes the 1 + 10 digit dial
string is required to make long distance calls through the PSTN, and that sevendigit
calling would use the PSTN for local calls. Even though the addition of the
www.syngress.com
312 Chapter 9 • AVVID Dial Plans
9.@ (as discussed earlier) includes the ability to dial 911 for emergencies, the Cisco
IOS gateway requires an entry for the dial peer.You could also add dial peers for
411 if you want that service to be available (this is also handled by the 9.@).
NOTE
Voice over IP points to layer three (routed) addresses, so it looks like IP
packet-based traffic as it traverses the network. POTS traffic (and VoFR)
is based on layer two, so it is treated as a voice call all the way to
completion.
Setting Up Single-Site Campuses
Setting Up Single-Site Campuses
In many instances, you will implement AVVID-based solutions in a single site
configuration.These are the implementations that only have one office and no
www.syngress.com
310 Chapter 9 • AVVID Dial Plans
WAN connections to external sites.When configuring these types of sites, you
will often implement a simple dial plan that can provide intraoffice calling (with
four or five digits depending on the site), as well as connections to the PSTN
(usually by dialing a 9). Long distance would also be handled by the PSTN, with
the dialing party using a 9, then a 1, followed by the area code before dialing the
seven-digit number. If you plan to use multiple carries for your PSTN, you may
have a scenario that flows like that in Figure 9.4.
In many instances, you will implement AVVID-based solutions in a single site
configuration.These are the implementations that only have one office and no
www.syngress.com
310 Chapter 9 • AVVID Dial Plans
WAN connections to external sites.When configuring these types of sites, you
will often implement a simple dial plan that can provide intraoffice calling (with
four or five digits depending on the site), as well as connections to the PSTN
(usually by dialing a 9). Long distance would also be handled by the PSTN, with
the dialing party using a 9, then a 1, followed by the area code before dialing the
seven-digit number. If you plan to use multiple carries for your PSTN, you may
have a scenario that flows like that in Figure 9.4.
Guidelines for the Design and Implementation of Dial Plans
Guidelines for the Design and
Implementation of Dial Plans
As with any project, its complexity will depend on the number of variables factored
in. Dial plan complexity can vary, based on any number of configuration
choices, such as the total amount of paths a call can be sent through.What I will
do in the following section is try to give you an idea of what to expect with
some of the usual dial plan implementations.
Implementation of Dial Plans
As with any project, its complexity will depend on the number of variables factored
in. Dial plan complexity can vary, based on any number of configuration
choices, such as the total amount of paths a call can be sent through.What I will
do in the following section is try to give you an idea of what to expect with
some of the usual dial plan implementations.
Creating a Calling Search Space
Creating a Calling Search Space
What is a calling search space? It is a list of partitions that can be accessed by
users so they can place a call.These calling search spaces are only allocated to
telephony devices that can start calls. Once implemented, it is simple to create
and use dialing restrictions because users are only allowed to dial those partitions
in the calling search space they are assigned to. If the user tries to call outside the
allowed partitions, they receive a busy signal.
For all intents and purposes, the calling search space is what allows callers to
complete connections for their calls.You would often use this configuration
when setting up office call policy. For example, when you set up office phones,
you often allow them unrestricted dialing abilities. Lobby phones, on the other
hand, can usually only call other phones located in the office.To establish these
criteria, you must create a partition for the office users, in this case SE-Users (see
Table 9.6). All calls destined for the PSTN would have the route pattern 9, and
those calls would be placed within the SE-PSTN allocated partition.Two calling
search spaces would then need to be created to represent the two sets of dialing
characteristics (see Table 9.7).
What is a calling search space? It is a list of partitions that can be accessed by
users so they can place a call.These calling search spaces are only allocated to
telephony devices that can start calls. Once implemented, it is simple to create
and use dialing restrictions because users are only allowed to dial those partitions
in the calling search space they are assigned to. If the user tries to call outside the
allowed partitions, they receive a busy signal.
For all intents and purposes, the calling search space is what allows callers to
complete connections for their calls.You would often use this configuration
when setting up office call policy. For example, when you set up office phones,
you often allow them unrestricted dialing abilities. Lobby phones, on the other
hand, can usually only call other phones located in the office.To establish these
criteria, you must create a partition for the office users, in this case SE-Users (see
Table 9.6). All calls destined for the PSTN would have the route pattern 9, and
those calls would be placed within the SE-PSTN allocated partition.Two calling
search spaces would then need to be created to represent the two sets of dialing
characteristics (see Table 9.7).
The Assignment of Partitions Partition Name Devices Designated to Partition
The Assignment of Partitions
Partition Name Devices Designated to Partition
SE-Users All office telephony devices
SE-PSTN All devices with routes destined for the PSTN
Table 9.7 The Assignment of Calling Search Space
Calling Search Space Partitions Devices Assigned
Unrestricted SE-Users All telephony devices
SE-PSTN that can make calls
SE-Internal SE-Users Telephony devices that cannot
call outside the local office
One of these calling search spaces would be labeled Unrestricted (to denote
the lack of restrictions on the calling device).This calling search space would
then have SE-Users and SE-PSTN associated with it.The second calling search
space (called SE-Internal) would then have only SE-Users associated with it.
Office users in the Unrestricted calling search space will be allowed to dial
anywhere, while telephony devices associated with the SE-Internal calling search
space will only be allowed to call internally.
From this basic configuration, you could add all sorts of calling features,
depending on the needs of your office.These include:
Limiting telephony devices to intrasite (local office) calling
Limiting telephony devices to intrasite calling, with emergency
calling ability (emergency calling is required for most, if not all, office
configurations)
Intrasite and intersite (external offices) calling
Intrasite and intersite calling, with emergency calling capability
Intrasite, intersite, emergency, and local PSTN calling
Intrasite, intersite, emergency, and national PSTN calling
Unrestricted calling (includes all the preceding, plus international calling)
These partitions and calling search spaces can allow autonomous dial ranges
on a partition basis. Extension and access codes located within different partitions
www.syngress.com
AVVID Dial Plans • Chapter 9 309
can have overlapping number schemes, and still work independently of each
other.This is usually seen in the implementation of a centralized call processing
system. In this example, all sites that use the same Cisco CallManager can dial the
number 9 to access the PSTN, even if they are located on different WAN segments.
When using a centralized Cisco CallManager for call processing, certain conditions
apply to overlapping users and extensions located at other sites.These
include:
Overlapping internal dial plans are supported if there is an implementation
of, or need for, voice mail on those extensions.This prevents issues
with the Cisco CallManager sending calls to voice mail and having to
decide which partition the call is destined for.The Cisco CallManager is
not designed to be intuitive, so a call directed to ext. 3637 in Seattle
cannot be distinguished from a call directed to ext. 3637 in San
Francisco.Voice mail requires unique extensions for identification.
If you do not require voice mail, you can have multiple sites with the
same extension.These extensions can be reached via:
The PSTN, dial the area code (if necessary), local access (exchange)
code (747), followed by the full directory number (3637).
The WAN, through the implementation of translation tables.The
tables can allow prepending of a unique code (sometimes referred to
as a steering code) to occur on extensions that overlap.This steering
code is then removed from the call when the destination is reached.
Partition Name Devices Designated to Partition
SE-Users All office telephony devices
SE-PSTN All devices with routes destined for the PSTN
Table 9.7 The Assignment of Calling Search Space
Calling Search Space Partitions Devices Assigned
Unrestricted SE-Users All telephony devices
SE-PSTN that can make calls
SE-Internal SE-Users Telephony devices that cannot
call outside the local office
One of these calling search spaces would be labeled Unrestricted (to denote
the lack of restrictions on the calling device).This calling search space would
then have SE-Users and SE-PSTN associated with it.The second calling search
space (called SE-Internal) would then have only SE-Users associated with it.
Office users in the Unrestricted calling search space will be allowed to dial
anywhere, while telephony devices associated with the SE-Internal calling search
space will only be allowed to call internally.
From this basic configuration, you could add all sorts of calling features,
depending on the needs of your office.These include:
Limiting telephony devices to intrasite (local office) calling
Limiting telephony devices to intrasite calling, with emergency
calling ability (emergency calling is required for most, if not all, office
configurations)
Intrasite and intersite (external offices) calling
Intrasite and intersite calling, with emergency calling capability
Intrasite, intersite, emergency, and local PSTN calling
Intrasite, intersite, emergency, and national PSTN calling
Unrestricted calling (includes all the preceding, plus international calling)
These partitions and calling search spaces can allow autonomous dial ranges
on a partition basis. Extension and access codes located within different partitions
www.syngress.com
AVVID Dial Plans • Chapter 9 309
can have overlapping number schemes, and still work independently of each
other.This is usually seen in the implementation of a centralized call processing
system. In this example, all sites that use the same Cisco CallManager can dial the
number 9 to access the PSTN, even if they are located on different WAN segments.
When using a centralized Cisco CallManager for call processing, certain conditions
apply to overlapping users and extensions located at other sites.These
include:
Overlapping internal dial plans are supported if there is an implementation
of, or need for, voice mail on those extensions.This prevents issues
with the Cisco CallManager sending calls to voice mail and having to
decide which partition the call is destined for.The Cisco CallManager is
not designed to be intuitive, so a call directed to ext. 3637 in Seattle
cannot be distinguished from a call directed to ext. 3637 in San
Francisco.Voice mail requires unique extensions for identification.
If you do not require voice mail, you can have multiple sites with the
same extension.These extensions can be reached via:
The PSTN, dial the area code (if necessary), local access (exchange)
code (747), followed by the full directory number (3637).
The WAN, through the implementation of translation tables.The
tables can allow prepending of a unique code (sometimes referred to
as a steering code) to occur on extensions that overlap.This steering
code is then removed from the call when the destination is reached.
Designing a Dial Plan to Meet Your Needs
Designing a Dial Plan to Meet Your Needs
Designing a dial plan that meets your needs sounds pretty fundamental, but what
does it mean? When you implement AVVID, you should work under the assumption
that the less complex it is, the better. Find out what is used on a normal
(daily) basis, and what features are seldom used.With these answers, you can
create a plan that meets the needs of the client.
If you are setting up a branch office, you will probably need to implement a
system similar to this. Company X would like to set up AVVID within their
regional offices. All branch offices will have several levels of call barring that
allow local calls (those calls located within the local exchange only), some that
allow long distance calls, and some that allow international calling. For ease of
demonstration, we will create dial plans that allow for a greater level of granularity
then you might encounter in the real world. By creating a high level of distinction,
you will be able to filter numbers using the local dialing prefix from all
other number combinations.This will help place these router patterns into separate
partitions and calling search spaces (as discussed earlier).This setup allows for
the control of end telephony devices and their ability for outdial access.
www.syngress.com
AVVID Dial Plans • Chapter 9 313
NOTE
You will need to alter the dial plan to fit the local numbers where the
CallManager is located. The [ ] wildcards will allow you to specify a range
of numbers, which should reduce the overall number of route patterns
that are comparable.
The United States (and North American) standard for dialing is based on
seven digits for the local exchange area, three digits for the area code, and a
leading one (1) for long distance calls (1 + 425 + (555-5555)).Table 9.8 shows a
basic dial plan.
NOTE
Table 9.8 is not an exhaustive list of all possible call combinations. It is
quite possible there are other numbers that aren’t listed, so please investigate
the particular dial plans associated with your location. Phone
books often have lists of area code and informational/service numbers.
Designing a dial plan that meets your needs sounds pretty fundamental, but what
does it mean? When you implement AVVID, you should work under the assumption
that the less complex it is, the better. Find out what is used on a normal
(daily) basis, and what features are seldom used.With these answers, you can
create a plan that meets the needs of the client.
If you are setting up a branch office, you will probably need to implement a
system similar to this. Company X would like to set up AVVID within their
regional offices. All branch offices will have several levels of call barring that
allow local calls (those calls located within the local exchange only), some that
allow long distance calls, and some that allow international calling. For ease of
demonstration, we will create dial plans that allow for a greater level of granularity
then you might encounter in the real world. By creating a high level of distinction,
you will be able to filter numbers using the local dialing prefix from all
other number combinations.This will help place these router patterns into separate
partitions and calling search spaces (as discussed earlier).This setup allows for
the control of end telephony devices and their ability for outdial access.
www.syngress.com
AVVID Dial Plans • Chapter 9 313
NOTE
You will need to alter the dial plan to fit the local numbers where the
CallManager is located. The [ ] wildcards will allow you to specify a range
of numbers, which should reduce the overall number of route patterns
that are comparable.
The United States (and North American) standard for dialing is based on
seven digits for the local exchange area, three digits for the area code, and a
leading one (1) for long distance calls (1 + 425 + (555-5555)).Table 9.8 shows a
basic dial plan.
NOTE
Table 9.8 is not an exhaustive list of all possible call combinations. It is
quite possible there are other numbers that aren’t listed, so please investigate
the particular dial plans associated with your location. Phone
books often have lists of area code and informational/service numbers.
Partitioning with Cisco CallManager
Partitioning with Cisco CallManager
So what is a partition? A partition is a group of telephony devices that have similar
reach ability.These devices are composed of route patterns, IP SoftPhones,
directory numbers, and so on.When creating partitions spaces, it’s a good idea to
group together those with similar characteristics and give them a name that
reflects those qualities. For example, if you have your System Engineers in
building A, North then you should create a group name something like SE-AN.
So what is a partition? A partition is a group of telephony devices that have similar
reach ability.These devices are composed of route patterns, IP SoftPhones,
directory numbers, and so on.When creating partitions spaces, it’s a good idea to
group together those with similar characteristics and give them a name that
reflects those qualities. For example, if you have your System Engineers in
building A, North then you should create a group name something like SE-AN.
Creation of Calling Restrictions and Configuration of Dial Plan Groups
Creation of Calling Restrictions and
Configuration of Dial Plan Groups
Within Cisco CallManager, you can create calling restrictions on a per telephony
device, or create closed dial plan groups (as long as they fall within the same
Cisco CallManager).What this means is that users that reside within the same
Cisco CallManager can be grouped together with the same calling restrictions
and dial plans. For example, if you have development teams that only need to talk
to each other, you can restrict their dial-plans to within the group, or limit their
www.syngress.com
AVVID Dial Plans • Chapter 9 307
ability to call long distance.Within the same Cisco CallManager, you may have
Accounting or Human Resources that need to make more long distance calls, so
you create calling communities based on their need.
These different communities are able to operate independently and can all
share the same gateway since they have overlapping dial-plans.You will find this
more useful in sites linked across WAN connections if they all share a central
Cisco CallManager as the call processing area.This also allows for the usage of
partitions and calling search space within the organization.
Configuration of Dial Plan Groups
Within Cisco CallManager, you can create calling restrictions on a per telephony
device, or create closed dial plan groups (as long as they fall within the same
Cisco CallManager).What this means is that users that reside within the same
Cisco CallManager can be grouped together with the same calling restrictions
and dial plans. For example, if you have development teams that only need to talk
to each other, you can restrict their dial-plans to within the group, or limit their
www.syngress.com
AVVID Dial Plans • Chapter 9 307
ability to call long distance.Within the same Cisco CallManager, you may have
Accounting or Human Resources that need to make more long distance calls, so
you create calling communities based on their need.
These different communities are able to operate independently and can all
share the same gateway since they have overlapping dial-plans.You will find this
more useful in sites linked across WAN connections if they all share a central
Cisco CallManager as the call processing area.This also allows for the usage of
partitions and calling search space within the organization.
Matching Variable-Length Dial Peers
Matching Variable-Length Dial Peers
Routers are configured by default to match variable-length dial peers. As long as
the digits dialed match the pattern on the dial peer, it will continue to filter.
Once you are processing digits beyond the matching point, however, the router
will ignore them during the filtering process. For example, the dial string for
information, 5551212, would be properly matched with the following dial peers:
dial-peer voice 1 voip
destination-pattern 555
session target ipv4:10.1.100.1
dial-peer voice 2 voip
destination-pattern 5551212
session target ipv4:10.1.100.2
In order to disable the matching of variable-length dial peers, you would add
the $ character at the end of the destination-pattern.The $ character will stop the
dial peer from matching the digits that would come after it, even if they were
able to be processed by another destination-pattern, as in the following example:
dial-peer voice 1 voip
destination-pattern 555$
session target ipv4:10.1.100.1
dial-peer voice 2 voip
destination-pattern 5551212
session target ipv4:10.1.100.2
With the $ at the end of the destination pattern, the dial peer for 5551212
would not be matched.The pattern would only match up to the 555 configured
for dial peer 1.
www.syngress.com
306 Chapter 9 • AVVID Dial Plans
As noted earlier, two-stage dialing collects digits that are dialed. It actually
collects them one by one and will attempt to match a dial peer after each digit is
dialed and processed. Once a match is found, the call will be routed. So, dialing
5551212 and using the following configuration:
dial-peer voice 1 voip
destination-pattern 555
session target ipv4:10.1.100.1
dial-peer voice 2 voip
destination-pattern 5551212
session target ipv4:10.1.100.2
you would see that the router would match the digits immediately to dial peer 1
and route the call.
In order for the digits to match the second dial peer, you would need to use
the timeout character (T) at the end of the destination pattern, in this case 555.
This would allow the digits a time limit with which to dial all numbers, and that
would allow the pattern to be matched to the best fit.This configuration would
look something like this:
dial-peer voice 1 voip
destination-pattern 555T
session target ipv4:10.1.100.1
dial-peer voice 2 voip
destination-pattern 5551212T
session target ipv4:10.1.100.2
Be aware that the router will also select dial peers based on whether the call
is inbound or outbound.
Routers are configured by default to match variable-length dial peers. As long as
the digits dialed match the pattern on the dial peer, it will continue to filter.
Once you are processing digits beyond the matching point, however, the router
will ignore them during the filtering process. For example, the dial string for
information, 5551212, would be properly matched with the following dial peers:
dial-peer voice 1 voip
destination-pattern 555
session target ipv4:10.1.100.1
dial-peer voice 2 voip
destination-pattern 5551212
session target ipv4:10.1.100.2
In order to disable the matching of variable-length dial peers, you would add
the $ character at the end of the destination-pattern.The $ character will stop the
dial peer from matching the digits that would come after it, even if they were
able to be processed by another destination-pattern, as in the following example:
dial-peer voice 1 voip
destination-pattern 555$
session target ipv4:10.1.100.1
dial-peer voice 2 voip
destination-pattern 5551212
session target ipv4:10.1.100.2
With the $ at the end of the destination pattern, the dial peer for 5551212
would not be matched.The pattern would only match up to the 555 configured
for dial peer 1.
www.syngress.com
306 Chapter 9 • AVVID Dial Plans
As noted earlier, two-stage dialing collects digits that are dialed. It actually
collects them one by one and will attempt to match a dial peer after each digit is
dialed and processed. Once a match is found, the call will be routed. So, dialing
5551212 and using the following configuration:
dial-peer voice 1 voip
destination-pattern 555
session target ipv4:10.1.100.1
dial-peer voice 2 voip
destination-pattern 5551212
session target ipv4:10.1.100.2
you would see that the router would match the digits immediately to dial peer 1
and route the call.
In order for the digits to match the second dial peer, you would need to use
the timeout character (T) at the end of the destination pattern, in this case 555.
This would allow the digits a time limit with which to dial all numbers, and that
would allow the pattern to be matched to the best fit.This configuration would
look something like this:
dial-peer voice 1 voip
destination-pattern 555T
session target ipv4:10.1.100.1
dial-peer voice 2 voip
destination-pattern 5551212T
session target ipv4:10.1.100.2
Be aware that the router will also select dial peers based on whether the call
is inbound or outbound.
What Is Two-Stage Dialing?
What Is Two-Stage Dialing?
When a voice call is destined for the network, the router placing the call collects
all the dialed digits. It then takes these digits and filters them through the dial
peers to see if there is a match. Once a match is found, the router then immediately
places the call by forwarding the dialed string. Once the call is forwarded,
the router no longer collects digits for that session and they are dropped. Digits
and wildcards used in the destination pattern choose how many digits the router
collects before it tries to filter them through the dial peers.
When a voice call is destined for the network, the router placing the call collects
all the dialed digits. It then takes these digits and filters them through the dial
peers to see if there is a match. Once a match is found, the router then immediately
places the call by forwarding the dialed string. Once the call is forwarded,
the router no longer collects digits for that session and they are dropped. Digits
and wildcards used in the destination pattern choose how many digits the router
collects before it tries to filter them through the dial peers.
Options for External Calls Using Route Patterns
Options for External Calls Using Route Patterns
As discussed earlier, Cisco CallManager will wait ten seconds before
assuming dialing is completed. There are two options that can be added
to route patterns destined for outside North America through the PSTN.
The more common of the two is dialing the number zero (0).
To configure this option, you could add the statement:
Route Pattern = 0.!
0. is necessary to access the PSTN, while ! is the wildcard that represents
a digit (or number of digits). With this setup, the Cisco
CallManager still waits ten seconds to see if any more digits are dialed.
If none follow, the Cisco CallManager assumes the dialing is complete
and routes the call.
There is also the second option. This configuration instructs users
to dial a pound sign (#) to end the dial string so the call can be placed
immediately. The drawback is that you are expecting the user to listen to
the instruction and change their existing dialing habits. As you know,
people aren’t always happy with change, especially if they are used to
something easier (or that they are familiar with).
Route Pattern = 0.!#
0. is the code necessary to access the PSTN, while ! is the wildcard
that represents a digit (or number of digits). With this setup, the Cisco
CallManager will still wait ten seconds to see if any more digits are
dialed. If none follow, the Cisco CallManager assumes dialing is complete
and routes the call. The # (pound) is the end character. When the
user dials the pound key, the Cisco CallManager terminates the dialing
string and immediately routes the call.
As discussed earlier, Cisco CallManager will wait ten seconds before
assuming dialing is completed. There are two options that can be added
to route patterns destined for outside North America through the PSTN.
The more common of the two is dialing the number zero (0).
To configure this option, you could add the statement:
Route Pattern = 0.!
0. is necessary to access the PSTN, while ! is the wildcard that represents
a digit (or number of digits). With this setup, the Cisco
CallManager still waits ten seconds to see if any more digits are dialed.
If none follow, the Cisco CallManager assumes the dialing is complete
and routes the call.
There is also the second option. This configuration instructs users
to dial a pound sign (#) to end the dial string so the call can be placed
immediately. The drawback is that you are expecting the user to listen to
the instruction and change their existing dialing habits. As you know,
people aren’t always happy with change, especially if they are used to
something easier (or that they are familiar with).
Route Pattern = 0.!#
0. is the code necessary to access the PSTN, while ! is the wildcard
that represents a digit (or number of digits). With this setup, the Cisco
CallManager will still wait ten seconds to see if any more digits are
dialed. If none follow, the Cisco CallManager assumes dialing is complete
and routes the call. The # (pound) is the end character. When the
user dials the pound key, the Cisco CallManager terminates the dialing
string and immediately routes the call.
Longest Match Translation
Longest Match Translation
Cisco CallManager is also able to handle the longest match criteria with the
implementation of wildcard masks. For example, if there is a phone with a DID
located within the 0000–0999 range, the Cisco CallManager will direct the call
to that specific phone. In instances where there is no matching extension, then
the call will be matched against the translation table, and (using the previous
example) be routed to the front desk at 0001.
Digits can also be manipulated within the route pattern configuration using
Called/Calling Party Transformation, a method which allows for three types of
translations to occur within the called-number.These are:
The removal of digits from the dialed string
Application of the Transformation Mask to the called-party
The ability to prefix digits to the dialed string
These translations can be helpful in companies that either have a lot of
unused numbers or companies have multiple numbers. For example, a Cisco
CallManager may have a defined route pattern of 8XXXX to route a call to
another company office.The number being called is 0000, and needs to go
through the PBX.The route pattern has a called party number of 536XXX and a
calling party number of 15108XXXX.The calling party information mask of
1510 will be prepended to the calling-number.The access code (the number 8)
will be discarded from the dial string, and the digits 536 will be prefixed to the
number. All this will happen in that order so it can be properly translated to the
internal calling-number of 1510536XXXX so it can be routed by the internal
PBX.
The other way to do this is to use the called-party transformation mask of
536XXXX.The drawback to this is that the calling party transformation mask
www.syngress.com
302 Chapter 9 • AVVID Dial Plans
only applies to the calling-numbers, and the other transformation masks will only
apply to called-numbers. As noted earlier, the order of precedence for the Cisco
CallManager will be to remove digits from the dialed string, then apply the transformation
mask to the called-party, and afterward prefix digits to the dialed
string.
Cisco CallManager is also able to handle the longest match criteria with the
implementation of wildcard masks. For example, if there is a phone with a DID
located within the 0000–0999 range, the Cisco CallManager will direct the call
to that specific phone. In instances where there is no matching extension, then
the call will be matched against the translation table, and (using the previous
example) be routed to the front desk at 0001.
Digits can also be manipulated within the route pattern configuration using
Called/Calling Party Transformation, a method which allows for three types of
translations to occur within the called-number.These are:
The removal of digits from the dialed string
Application of the Transformation Mask to the called-party
The ability to prefix digits to the dialed string
These translations can be helpful in companies that either have a lot of
unused numbers or companies have multiple numbers. For example, a Cisco
CallManager may have a defined route pattern of 8XXXX to route a call to
another company office.The number being called is 0000, and needs to go
through the PBX.The route pattern has a called party number of 536XXX and a
calling party number of 15108XXXX.The calling party information mask of
1510 will be prepended to the calling-number.The access code (the number 8)
will be discarded from the dial string, and the digits 536 will be prefixed to the
number. All this will happen in that order so it can be properly translated to the
internal calling-number of 1510536XXXX so it can be routed by the internal
PBX.
The other way to do this is to use the called-party transformation mask of
536XXXX.The drawback to this is that the calling party transformation mask
www.syngress.com
302 Chapter 9 • AVVID Dial Plans
only applies to the calling-numbers, and the other transformation masks will only
apply to called-numbers. As noted earlier, the order of precedence for the Cisco
CallManager will be to remove digits from the dialed string, then apply the transformation
mask to the called-party, and afterward prefix digits to the dialed
string.
Digit Translation Tables
Digit Translation Tables
The ability to manipulate dialed digits is supported within Cisco Call Manager.
What this allows for is the manipulation of not only the digits themselves, but
also the number of digits within the string.This is most commonly seen in the
www.syngress.com
The Usage of H.323 Gateways
A device that is “gateway controlled” will need to successfully query the
gatekeeper in order to gain admission. The CODEC region should be set
to handle the correct CODEC and compression technique. It is allowable
to share H.323 gateways between multiple inbound and outbound calls.
Gateways that are implemented with Skinny Gateway Protocol and
MGCP are only allowed within one Cisco CallManager cluster.
Configuring & Implementing…
AVVID Dial Plans • Chapter 9 301
directing of calls that have no directly defined destination or DID number.These
calls are usually forwarded to an attendant or voice mail. As an example, if your
office uses the DID range of 0000 to 0999 and you want calls to be forwarded to
the front desk, which is defined as 0001, you can create a translation table of
0XXX with a translation mask of 0001.This will direct calls to the front desk
destination. Note:This is for DID numbers that are not defined.This also works
for hunt groups, and can be used for internal (also called on-net if within the
network) and external (also known as off-net) calls as well as for inbound and
outbound calls.
The ability to manipulate dialed digits is supported within Cisco Call Manager.
What this allows for is the manipulation of not only the digits themselves, but
also the number of digits within the string.This is most commonly seen in the
www.syngress.com
The Usage of H.323 Gateways
A device that is “gateway controlled” will need to successfully query the
gatekeeper in order to gain admission. The CODEC region should be set
to handle the correct CODEC and compression technique. It is allowable
to share H.323 gateways between multiple inbound and outbound calls.
Gateways that are implemented with Skinny Gateway Protocol and
MGCP are only allowed within one Cisco CallManager cluster.
Configuring & Implementing…
AVVID Dial Plans • Chapter 9 301
directing of calls that have no directly defined destination or DID number.These
calls are usually forwarded to an attendant or voice mail. As an example, if your
office uses the DID range of 0000 to 0999 and you want calls to be forwarded to
the front desk, which is defined as 0001, you can create a translation table of
0XXX with a translation mask of 0001.This will direct calls to the front desk
destination. Note:This is for DID numbers that are not defined.This also works
for hunt groups, and can be used for internal (also called on-net if within the
network) and external (also known as off-net) calls as well as for inbound and
outbound calls.
The Usage of H.323 Gateways
The Usage of H.323 Gateways
A device that is “gateway controlled” will need to successfully query the
gatekeeper in order to gain admission. The CODEC region should be set
to handle the correct CODEC and compression technique. It is allowable
to share H.323 gateways between multiple inbound and outbound calls.
Gateways that are implemented with Skinny Gateway Protocol and
MGCP are only allowed within one Cisco CallManager cluster.
A device that is “gateway controlled” will need to successfully query the
gatekeeper in order to gain admission. The CODEC region should be set
to handle the correct CODEC and compression technique. It is allowable
to share H.323 gateways between multiple inbound and outbound calls.
Gateways that are implemented with Skinny Gateway Protocol and
MGCP are only allowed within one Cisco CallManager cluster.
Fixed-Length Dial Peers versus Variable-Length Dial Peers
Fixed-Length Dial Peers versus
Variable-Length Dial Peers
When considering how to implement your Voice over network, you need to
think about the number of digits the router will be dealing with. If you only
have fixed-length dialing where users apply four or five-digit dialing to connect
to other office phones, the creation of dial plans is really quite simple.You need
to know the destination patterns used and build the dial peers based on destination
patterns.
On the other hand, some users will need to have full dialing privileges for all
their calling needs.When this is the case, you need to implement variable-length
dialing plans, something which is bit more complicated.When unsure about the
dialing habits of office users, you are generally left with two options:
You could create a dial plan that includes all possible prefixes and wild
cards to ensure all calls are routed (not fun).
You could implement variable-length dial peers.The router or Cisco
CallManager will then collect the dialed string digits and route them
based on pattern matching (highly recommended).
Remember that fixed-length peers are exactly that, fixed length.They will
always have the same number of digits associated with them whether they are
wildcard digits or just dialed digits. For example, if you only configure your
router or Cisco CallManager for fixed-length dial plans, the digits received by the
router (or Cisco CallManager) must have the appropriate number of dialed digits.
If you set up the router to accept ten-digit calls, the router will only connect
once all ten digits are dialed. If in this scenario you set up a static area code along
with seven digits, and the user doesn’t dial that area code, the call will not be able
to complete because it does not match the dial peer.
Variable-length dial plans allow for the router or Cisco CallManager to
receive inconsistent dialed digits and compare them to its routing table. It can do
this through the configuration of several options. For example there can be the
inclusion of the command destination-pattern with its options.The following
configuration of a variable-length dial peer will hopefully give you some idea of
what we’re talking about, and the explanation that follows should illuminate the
configuration.
dial-peer voice 1 voip \\Sets the dial peer as VoIP
destination-pattern 9T \\dial peer must be matched when the
\\router receives the Number 9 + any
www.syngress.com
304 Chapter 9 • AVVID Dial Plans
\\digits, or the call will terminate
session target ip4:10.1.100.1 \\When the dial peer is matched, it will
\\ setup a call to 10.1.100.1
Several characters, used as switches, can be inserted into the destinationpattern
command.The preceding configuration uses the “T” switch, which is a
timeout character.You could also configure a termination character defined with
the command dial peer terminator. I prefer to use “#”
for termination, but you can choose other characters. Keep in mind, though, that
this command can only be found on routers that are voice-enabled.To enable the
termination character, you can do any of the following:
Use the “#” as termination character that can be sent from the
Telephony device, thus making it like a Cell phone send key. So the
router would receive the “#” character and know that it need to send all
of the characters that were dialed before the “#” key.
When configuring the voice-port, you can add the command “timeouts
interdigit” and define the amount of time that router or Cisco
CallManager will wait between dialed digits (normally set to 10 seconds
by default) before sending the digits.You may want to configure this for
a smaller interval, as many times users will become inpatient with this
long a wait.
When you install Cisco CallManager within North America, you can use the
“@” character with the creation of route patterns to create variable length dial
plans.This way the user can dial a seven-digit local number, or ten- (area code +
number) and eleven- (1 + area code + number) digit numbers to call long distance.
When the number reaches the last dialed digit, the call will immediately be
placed.The “@” character will not work outside North America, though.
In order to construct variable-length dial plans in the past, you needed to
configure the Cisco CallManager with a router pattern that consisted of 0.!
within the setup. By setting up the wildcard, the Cisco CallManager would then
be able to utilize variable-length dial plans, but it also needed to use the timeout
after the last digit before it would place a call to the gateway.The alternative was
to create variable-length dial plans for the entire national calling-number scheme.
A lot of support issues needed to be addressed to make this feasible, but it
allowed a myriad of calling features and offered users a minimal wait.
For international calls, you will need to implement the wildcard setup, as
North American systems are not designed to match foreign exchanges.
Variable-Length Dial Peers
When considering how to implement your Voice over network, you need to
think about the number of digits the router will be dealing with. If you only
have fixed-length dialing where users apply four or five-digit dialing to connect
to other office phones, the creation of dial plans is really quite simple.You need
to know the destination patterns used and build the dial peers based on destination
patterns.
On the other hand, some users will need to have full dialing privileges for all
their calling needs.When this is the case, you need to implement variable-length
dialing plans, something which is bit more complicated.When unsure about the
dialing habits of office users, you are generally left with two options:
You could create a dial plan that includes all possible prefixes and wild
cards to ensure all calls are routed (not fun).
You could implement variable-length dial peers.The router or Cisco
CallManager will then collect the dialed string digits and route them
based on pattern matching (highly recommended).
Remember that fixed-length peers are exactly that, fixed length.They will
always have the same number of digits associated with them whether they are
wildcard digits or just dialed digits. For example, if you only configure your
router or Cisco CallManager for fixed-length dial plans, the digits received by the
router (or Cisco CallManager) must have the appropriate number of dialed digits.
If you set up the router to accept ten-digit calls, the router will only connect
once all ten digits are dialed. If in this scenario you set up a static area code along
with seven digits, and the user doesn’t dial that area code, the call will not be able
to complete because it does not match the dial peer.
Variable-length dial plans allow for the router or Cisco CallManager to
receive inconsistent dialed digits and compare them to its routing table. It can do
this through the configuration of several options. For example there can be the
inclusion of the command destination-pattern with its options.The following
configuration of a variable-length dial peer will hopefully give you some idea of
what we’re talking about, and the explanation that follows should illuminate the
configuration.
dial-peer voice 1 voip \\Sets the dial peer as VoIP
destination-pattern 9T \\dial peer must be matched when the
\\router receives the Number 9 + any
www.syngress.com
304 Chapter 9 • AVVID Dial Plans
\\digits, or the call will terminate
session target ip4:10.1.100.1 \\When the dial peer is matched, it will
\\ setup a call to 10.1.100.1
Several characters, used as switches, can be inserted into the destinationpattern
command.The preceding configuration uses the “T” switch, which is a
timeout character.You could also configure a termination character defined with
the command dial peer terminator
for termination, but you can choose other characters. Keep in mind, though, that
this command can only be found on routers that are voice-enabled.To enable the
termination character, you can do any of the following:
Use the “#” as termination character that can be sent from the
Telephony device, thus making it like a Cell phone send key. So the
router would receive the “#” character and know that it need to send all
of the characters that were dialed before the “#” key.
When configuring the voice-port, you can add the command “timeouts
interdigit” and define the amount of time that router or Cisco
CallManager will wait between dialed digits (normally set to 10 seconds
by default) before sending the digits.You may want to configure this for
a smaller interval, as many times users will become inpatient with this
long a wait.
When you install Cisco CallManager within North America, you can use the
“@” character with the creation of route patterns to create variable length dial
plans.This way the user can dial a seven-digit local number, or ten- (area code +
number) and eleven- (1 + area code + number) digit numbers to call long distance.
When the number reaches the last dialed digit, the call will immediately be
placed.The “@” character will not work outside North America, though.
In order to construct variable-length dial plans in the past, you needed to
configure the Cisco CallManager with a router pattern that consisted of 0.!
within the setup. By setting up the wildcard, the Cisco CallManager would then
be able to utilize variable-length dial plans, but it also needed to use the timeout
after the last digit before it would place a call to the gateway.The alternative was
to create variable-length dial plans for the entire national calling-number scheme.
A lot of support issues needed to be addressed to make this feasible, but it
allowed a myriad of calling features and offered users a minimal wait.
For international calls, you will need to implement the wildcard setup, as
North American systems are not designed to match foreign exchanges.
Telephony Devices
Telephony Devices
Any IP end device that can be entered into a route group can be considered
a telephony device. For example, a device that is configured to use H.323
gateways, such as an IP SoftPhones or Microsoft NetMeeting can be considered
a telephony device.
The route pattern dialing structures are usually used to connect IP phone
calls destined for external gateways or external Cisco CallManagers using H.323.
What this allows for is the ability to use alternate paths if the primary is unable
to accept or admit calls. For example, intraoffice calls that use WAN connections
as the primary path and the PSTN as the secondary path can choose the secondary
path to complete the call if the WAN is saturated. On the other hand,
devices that reside on the same Cisco CallManager are unable to use alternate
routes, so if there is a problem within the LAN, the phones are unable to reroute
to the PSTN to complete the call.
Any IP end device that can be entered into a route group can be considered
a telephony device. For example, a device that is configured to use H.323
gateways, such as an IP SoftPhones or Microsoft NetMeeting can be considered
a telephony device.
The route pattern dialing structures are usually used to connect IP phone
calls destined for external gateways or external Cisco CallManagers using H.323.
What this allows for is the ability to use alternate paths if the primary is unable
to accept or admit calls. For example, intraoffice calls that use WAN connections
as the primary path and the PSTN as the secondary path can choose the secondary
path to complete the call if the WAN is saturated. On the other hand,
devices that reside on the same Cisco CallManager are unable to use alternate
routes, so if there is a problem within the LAN, the phones are unable to reroute
to the PSTN to complete the call.
Route Groups
Route Groups
In order to control telephony devices like gateways, you create route groups.
These gateways can be created using H.323, MGCP, or Skinny Gateway Protocol.
End telephony devices that would use H.323 would be programs such as
Microsoft NetMeeting and the Cisco CallManager Remote Connections that act
as H.323 Gateways. In this setup, the route group can connect to one or more
devices, and is able to select between these devices based on preference. In this
instance, the route groups can direct all calls destined to the primary device to
the secondary device if the main device is not available. Again, this can be considered
a trunk group.
www.syngress.com
Digit Manipulation for the Cisco CallManager
You can only apply digit manipulation to route patterns for outbound
calls only. This is because the digits need to be sent to the route list plus
the route groups. Individual route groups can have specific digit
changes for the same route pattern. You usually see this where a dialed
number needs to have different modifications like when devices need to
dial seven digits to reach a remote office that has a four digit internal
dial plan. This often happens when you have a call that cannot be completed
through the WAN and needs to be routed though the PSTN. What
would occur is the Cisco CallManager would prepend the first three
digits onto the dial string. A route pattern can be associated with only
one route list.
Configuring & Implementing…
300 Chapter 9 • AVVID Dial Plans
You can also point one or more route lists to the same route group. All the
devices within this route group have the same characteristics, like path and dial
string changes.There is also prioritization, as the string manipulation in a route
group overrides the changes of a route pattern.
In order to control telephony devices like gateways, you create route groups.
These gateways can be created using H.323, MGCP, or Skinny Gateway Protocol.
End telephony devices that would use H.323 would be programs such as
Microsoft NetMeeting and the Cisco CallManager Remote Connections that act
as H.323 Gateways. In this setup, the route group can connect to one or more
devices, and is able to select between these devices based on preference. In this
instance, the route groups can direct all calls destined to the primary device to
the secondary device if the main device is not available. Again, this can be considered
a trunk group.
www.syngress.com
Digit Manipulation for the Cisco CallManager
You can only apply digit manipulation to route patterns for outbound
calls only. This is because the digits need to be sent to the route list plus
the route groups. Individual route groups can have specific digit
changes for the same route pattern. You usually see this where a dialed
number needs to have different modifications like when devices need to
dial seven digits to reach a remote office that has a four digit internal
dial plan. This often happens when you have a call that cannot be completed
through the WAN and needs to be routed though the PSTN. What
would occur is the Cisco CallManager would prepend the first three
digits onto the dial string. A route pattern can be associated with only
one route list.
Configuring & Implementing…
300 Chapter 9 • AVVID Dial Plans
You can also point one or more route lists to the same route group. All the
devices within this route group have the same characteristics, like path and dial
string changes.There is also prioritization, as the string manipulation in a route
group overrides the changes of a route pattern.
Route List
Route List
A route list is used to route a call. It is configured to map the routes of a call to
one or more route groups, which basically act as trunk groups.The route list will
then forward the call to the route group based on some predefined preferences.
For example, the main (primary) route group may be configured to route calls
based on cost and metrics, whereas the secondary route may be configured to
only be used in instances where the primary circuit is unavailable, like in an alltrunks-
busy condition when there isn’t enough bandwidth to admit or complete
a call.
A route list is used to route a call. It is configured to map the routes of a call to
one or more route groups, which basically act as trunk groups.The route list will
then forward the call to the route group based on some predefined preferences.
For example, the main (primary) route group may be configured to route calls
based on cost and metrics, whereas the secondary route may be configured to
only be used in instances where the primary circuit is unavailable, like in an alltrunks-
busy condition when there isn’t enough bandwidth to admit or complete
a call.
Digit Manipulation for the Cisco CallManager
Digit Manipulation for the Cisco CallManager
You can only apply digit manipulation to route patterns for outbound
calls only. This is because the digits need to be sent to the route list plus
the route groups. Individual route groups can have specific digit
changes for the same route pattern. You usually see this where a dialed
number needs to have different modifications like when devices need to
dial seven digits to reach a remote office that has a four digit internal
dial plan. This often happens when you have a call that cannot be completed
through the WAN and needs to be routed though the PSTN. What
would occur is the Cisco CallManager would prepend the first three
digits onto the dial string. A route pattern can be associated with only
one route list.
You can only apply digit manipulation to route patterns for outbound
calls only. This is because the digits need to be sent to the route list plus
the route groups. Individual route groups can have specific digit
changes for the same route pattern. You usually see this where a dialed
number needs to have different modifications like when devices need to
dial seven digits to reach a remote office that has a four digit internal
dial plan. This often happens when you have a call that cannot be completed
through the WAN and needs to be routed though the PSTN. What
would occur is the Cisco CallManager would prepend the first three
digits onto the dial string. A route pattern can be associated with only
one route list.
Number Expansion
Number Expansion
Many larger offices use extension numbers to dial internally between users,
instead of the entire E.164 telephone number. Extensions can be defined as a
destination-pattern for a dial peer.This way the router will recognize the extension
number and will be able to translate it into the E.164 number; that is, if the
num-exp command has been implemented.
This will enable the router to prepend the digits you define before it passes
them to the remote telephony device.This will reduce the total number of digits
that must be dialed to complete a call to reach a user at a remote office location.
Number expansion is similar to implementing a prefix (discussed earlier), but
number expansion is applied to all dial peers, not just those defined.
An example of number expansion would be an office where you would dial
the last four digits of the E.164 address to reach someone within the company. In
this instance, the complete telephone number may be 747-3637, but internal
users would only have to dial 3637 to reach the particular user. All users located
at this office have the same first four digits (7473).With this information, you
cold configure the dial peers destination-patterns using each extension number,
and use number expansion to prepend those first four digits to the extension.The
router configurations would look like this:
num-exp 3… 7473…
dial peer voice 4 pots
destination pattern 3637
www.syngress.
Many larger offices use extension numbers to dial internally between users,
instead of the entire E.164 telephone number. Extensions can be defined as a
destination-pattern for a dial peer.This way the router will recognize the extension
number and will be able to translate it into the E.164 number; that is, if the
num-exp command has been implemented.
This will enable the router to prepend the digits you define before it passes
them to the remote telephony device.This will reduce the total number of digits
that must be dialed to complete a call to reach a user at a remote office location.
Number expansion is similar to implementing a prefix (discussed earlier), but
number expansion is applied to all dial peers, not just those defined.
An example of number expansion would be an office where you would dial
the last four digits of the E.164 address to reach someone within the company. In
this instance, the complete telephone number may be 747-3637, but internal
users would only have to dial 3637 to reach the particular user. All users located
at this office have the same first four digits (7473).With this information, you
cold configure the dial peers destination-patterns using each extension number,
and use number expansion to prepend those first four digits to the extension.The
router configurations would look like this:
num-exp 3… 7473…
dial peer voice 4 pots
destination pattern 3637
www.syngress.
What Is Digit Manipulation, and How Do You Configure It?
What Is Digit Manipulation, and How
Do You Configure It?
In the real world, you may have a device that is already in use, so why rock the
boat and change everything in one shot? You may find it easier to transition to
AVVID if you can also leverage your existing equipment, such as Key System
Units (KSU) and PBX equipment. As always, however, issues will arise, and you
may want to maintain certain added functionalities. For example, many PBXs can
accept dialed digits for the PSTN and international calls. So you may need to
configure digit manipulation within your dial peers so you can utilize your
current dial plans.
Do You Configure It?
In the real world, you may have a device that is already in use, so why rock the
boat and change everything in one shot? You may find it easier to transition to
AVVID if you can also leverage your existing equipment, such as Key System
Units (KSU) and PBX equipment. As always, however, issues will arise, and you
may want to maintain certain added functionalities. For example, many PBXs can
accept dialed digits for the PSTN and international calls. So you may need to
configure digit manipulation within your dial peers so you can utilize your
current dial plans.
Route Pattern-Flow of a Call through a Cisco CallManager Route Pattern
Flow of a Call through a Cisco CallManager Route Pattern
Route Pattern
(1)
Route List
(2)
Route Group
(4)
Route Group
(3)
Calling
Party
(A)
Destination
(B)
WAN
PSTN
AVVID Dial Plans • Chapter 9 297
Route Pattern
A route pattern is the addressing method that identifies the dialed number and
uses route lists and route group configurations to determine the route for call
completion. Dialed numbers (E.164 North American Standard) are broken down
into smaller groups, creating route patterns that can be entered into the Cisco
CallManager as a specific number (for point-to-point direct dialing) or as a
number range (the more common implementation). By using a route pattern,
you can summarize a large range of numbers so minimal entries are needed to
route a call.
As a dialed number is routed, the CallManager will look to create a pattern
match, so the call can be correctly routed to the next hop and eventually to the
end devices. Keep in mind, the digits can still be changed by the CallManager
before they are put into the route list. By this method, numbers can be added or
subtracted to the dialed strings. Once the number is passed to the route list, it
will determine which route it will take to its next route groups (also trunk
groups) and prioritize the traffic and connections.
Route Pattern
(1)
Route List
(2)
Route Group
(4)
Route Group
(3)
Calling
Party
(A)
Destination
(B)
WAN
PSTN
AVVID Dial Plans • Chapter 9 297
Route Pattern
A route pattern is the addressing method that identifies the dialed number and
uses route lists and route group configurations to determine the route for call
completion. Dialed numbers (E.164 North American Standard) are broken down
into smaller groups, creating route patterns that can be entered into the Cisco
CallManager as a specific number (for point-to-point direct dialing) or as a
number range (the more common implementation). By using a route pattern,
you can summarize a large range of numbers so minimal entries are needed to
route a call.
As a dialed number is routed, the CallManager will look to create a pattern
match, so the call can be correctly routed to the next hop and eventually to the
end devices. Keep in mind, the digits can still be changed by the CallManager
before they are put into the route list. By this method, numbers can be added or
subtracted to the dialed strings. Once the number is passed to the route list, it
will determine which route it will take to its next route groups (also trunk
groups) and prioritize the traffic and connections.
External Calls
External Calls
Configuring Cisco CallManager to complete external calls requires implementing
a route pattern.A route pattern is used to direct calls off network to a PSTN
gateway. Route patterns can also be used if there are Cisco CallManagers located
on a WAN-connected network.
Cisco CallManager dial plans are usually deployed in a tier system.This
system lets different routes handle dialed numbers.You can also manipulate dialed
strings, based on network requirements.This manipulation can either add or subtract
digits from the number dialed by the user so as to accommodate network
and gateway needs. Cisco CallManager can also create Trunk groups that will
handle redundancy and create better paths for least-cost routing. For example,
when using trunk groups, the system has the ability to choose an alternate route
to complete (or in some cases admit) calls if the trunks do not have sufficient
bandwidth to handle the call.This can be denoted (when creating the dial plan)
as a continuation of the rule that moves calls to the PSTN if WAN connections
are saturated.
In Figure 9.3, a call is placed from a telephony device (A). It is then matched
against the route pattern (1), where digit manipulation takes place. From here, the
call is forwarded to the route list (2).The route list adds the preference of connecting
the call over the WAN link. If the call is unable to be completed through
the WAN (because of insufficient resources or some other reason), then the call
will be forwarded to the PSTN. If the PSTN cannot complete the connection,
then the user will receive a busy signal (unless there is a third route configured).
From either the WAN or the PSTN, the call is forwarded to the destination party
(B). Again, this entire process should be transparent to the end user.
Configuring Cisco CallManager to complete external calls requires implementing
a route pattern.A route pattern is used to direct calls off network to a PSTN
gateway. Route patterns can also be used if there are Cisco CallManagers located
on a WAN-connected network.
Cisco CallManager dial plans are usually deployed in a tier system.This
system lets different routes handle dialed numbers.You can also manipulate dialed
strings, based on network requirements.This manipulation can either add or subtract
digits from the number dialed by the user so as to accommodate network
and gateway needs. Cisco CallManager can also create Trunk groups that will
handle redundancy and create better paths for least-cost routing. For example,
when using trunk groups, the system has the ability to choose an alternate route
to complete (or in some cases admit) calls if the trunks do not have sufficient
bandwidth to handle the call.This can be denoted (when creating the dial plan)
as a continuation of the rule that moves calls to the PSTN if WAN connections
are saturated.
In Figure 9.3, a call is placed from a telephony device (A). It is then matched
against the route pattern (1), where digit manipulation takes place. From here, the
call is forwarded to the route list (2).The route list adds the preference of connecting
the call over the WAN link. If the call is unable to be completed through
the WAN (because of insufficient resources or some other reason), then the call
will be forwarded to the PSTN. If the PSTN cannot complete the connection,
then the user will receive a busy signal (unless there is a third route configured).
From either the WAN or the PSTN, the call is forwarded to the destination party
(B). Again, this entire process should be transparent to the end user.
The Mobility of IP Devices
The Mobility of IP Devices
IP phones are not the only network devices that work with DN connection
properties. Cisco CallManager will also maintain the DN with Cisco
IP SoftPhones, and certain types of analog devices (such as phones and
facsimile machines) connected to gateways that use MGCP or the Skinny
Gateway Protocol.
IP phones are not the only network devices that work with DN connection
properties. Cisco CallManager will also maintain the DN with Cisco
IP SoftPhones, and certain types of analog devices (such as phones and
facsimile machines) connected to gateways that use MGCP or the Skinny
Gateway Protocol.
Digit Removal and Prefixes
Digit Removal and Prefixes
When a dial string is matched to an outbound POTS dial peer, the terminating
router will remove the left-justified digits that were explicit matches for the
destination-pattern.The leftover digits would then be forwarded to the telephony
device, like the PSTN or the PBX. Sometimes, the telephony interface will need
digits removed so they can support the existing dial plan. If this is the case, you
can use the command no digit-strip in the dial peer configuration.This command
will disable the removal of the digits, or you could use the prefix dial
www.syngress.com
298 Chapter 9 • AVVID Dial Plans
peer command, which will prepend digits to the dial string before they are
forwarded to the interface. Be aware that these commands only work in POTS
dial peers.
When a dial string is matched to an outbound POTS dial peer, the terminating
router will remove the left-justified digits that were explicit matches for the
destination-pattern.The leftover digits would then be forwarded to the telephony
device, like the PSTN or the PBX. Sometimes, the telephony interface will need
digits removed so they can support the existing dial plan. If this is the case, you
can use the command no digit-strip in the dial peer configuration.This command
will disable the removal of the digits, or you could use the prefix dial
www.syngress.com
298 Chapter 9 • AVVID Dial Plans
peer command, which will prepend digits to the dial string before they are
forwarded to the interface. Be aware that these commands only work in POTS
dial peers.
Internal Calls
Internal Calls
The creation of dial plans for internal calls to IP phones registered within a Cisco
CallManager cluster is very simple.When the phone is initially configured, it is
assigned a directory number (DN).This DN is maintained throughout the configured
life of the phone. For example, if the phone is used in an office where
your users move frequently within the LAN, their phones can be unplugged and
connected to a different network jack, yet maintain their connection properties
(DN).When the phone is reconnected, it will update the Cisco CallManager
with its new IP address.
The creation of dial plans for internal calls to IP phones registered within a Cisco
CallManager cluster is very simple.When the phone is initially configured, it is
assigned a directory number (DN).This DN is maintained throughout the configured
life of the phone. For example, if the phone is used in an office where
your users move frequently within the LAN, their phones can be unplugged and
connected to a different network jack, yet maintain their connection properties
(DN).When the phone is reconnected, it will update the Cisco CallManager
with its new IP address.
Dial Plan Preferences
Dial Plan Preferences
It is generally considered a good idea to create a dial plan that preferences
certain paths routed across the IP network. If this network
becomes unavailable, then calls should be routed across the PSTN. As
always, the process should be transparent to the user.
It is generally considered a good idea to create a dial plan that preferences
certain paths routed across the IP network. If this network
becomes unavailable, then calls should be routed across the PSTN. As
always, the process should be transparent to the user.
Simplicity and Redundancy
Simplicity and Redundancy
Voice-
Enabled
Router
Voice-
Enabled
Router
Main
Office
IP Phone
WAN
IP Phone
PSTN
Cisco CallManager
GateKeeper
(With Redundancy)
Branch
Office
Voice-
Enabled
Router
Voice-
Enabled
Router
Main
Office
IP Phone
WAN
IP Phone
PSTN
Cisco CallManager
GateKeeper
(With Redundancy)
Branch
Office
Cisco CallManager Dial Plans
Cisco CallManager Dial Plans
By using Cisco CallManager, you are able to allow for greater growth and functionality
within your network because it was designed to be integrated with
Cisco’s Internet Operating System (IOS) gateways.
Cisco CallManager dial plans are usually created to handle two types of calls,
internal and external:
Internal calls are those calls initiated and terminated on Cisco IP phones
that are included (registered) to the Cisco CallManager cluster.
www.syngress.com
294 Chapter 9 • AVVID Dial Plans
External calls are those calls passed through a PSTN gateway or a Cisco
CallManager that originate across a WAN connection.
Figure 9.2 is a network designed to handle calls destined for the WAN and
the PSTN. For this setup, voice calls would set the preference for the WAN and
would only be routed to the PSTN if the WAN were down or unavailable.This
routing takes place transparently to the user. In Figure 9.2, the Cisco
CallManager Gatekeeper is a router assigned to manage this specific task as a
gatekeeper.This router could also handle other items, but often it is best to have
the router taking care of just Gatekeeper functions.
By using Cisco CallManager, you are able to allow for greater growth and functionality
within your network because it was designed to be integrated with
Cisco’s Internet Operating System (IOS) gateways.
Cisco CallManager dial plans are usually created to handle two types of calls,
internal and external:
Internal calls are those calls initiated and terminated on Cisco IP phones
that are included (registered) to the Cisco CallManager cluster.
www.syngress.com
294 Chapter 9 • AVVID Dial Plans
External calls are those calls passed through a PSTN gateway or a Cisco
CallManager that originate across a WAN connection.
Figure 9.2 is a network designed to handle calls destined for the WAN and
the PSTN. For this setup, voice calls would set the preference for the WAN and
would only be routed to the PSTN if the WAN were down or unavailable.This
routing takes place transparently to the user. In Figure 9.2, the Cisco
CallManager Gatekeeper is a router assigned to manage this specific task as a
gatekeeper.This router could also handle other items, but often it is best to have
the router taking care of just Gatekeeper functions.
Routing Outbound Calls through the PSTN
Routing Outbound Calls through the PSTN
Calls destined to be routed through the PSTN usually require only one route
pattern. In some offices, you may find it necessary to create an access code to
access the PSTN, such as dialing a 9 before the number. In North America, the
dialing convention is divided into sections.There is an area code (510), the
exchange number (536), and the station ID number (XXXX). In order to make a
long distance call (a call outside your calling area), you may also need to dial a 1
at the beginning of the string. In some cities the convention for ten-digit dialing
is always necessary to complete calls. In these circumstances it is necessary to dial
the area code, but not the preceding 1.
With Cisco CallManager, you are able to create route patterns allowing you
to route calls that differentiate between a local call that requires ten-digit dialing
and a call that only requires seven-digit dialing. If the rule is not set, then Cisco
CallManager will wait ten seconds without dialed digit detection, and will
assume if there are no other digits dialed, then the user has completed dialing.
Creation of a local PSTN gateway dial plan is easy (and mostly painless).
Gateways that are based on Skinny Gateway Protocol and MGCP will have their
dial plan information configured within Cisco CallManager itself, whereas H.323
gateways will require only a small set of dial peers.The dial peers are then used
by the gateway to direct calls destined for the PSTN through the Cisco
CallManager.
If you are located outside North America, the numbers of digits that must be
dialed for call completion differ. In this case, you will need to create multiple
length dial-plans.The problem is, with the current version of Cisco CallManager,
the system doesn’t know when the dialing is complete, so you need to create
specific route patterns.
Calls destined to be routed through the PSTN usually require only one route
pattern. In some offices, you may find it necessary to create an access code to
access the PSTN, such as dialing a 9 before the number. In North America, the
dialing convention is divided into sections.There is an area code (510), the
exchange number (536), and the station ID number (XXXX). In order to make a
long distance call (a call outside your calling area), you may also need to dial a 1
at the beginning of the string. In some cities the convention for ten-digit dialing
is always necessary to complete calls. In these circumstances it is necessary to dial
the area code, but not the preceding 1.
With Cisco CallManager, you are able to create route patterns allowing you
to route calls that differentiate between a local call that requires ten-digit dialing
and a call that only requires seven-digit dialing. If the rule is not set, then Cisco
CallManager will wait ten seconds without dialed digit detection, and will
assume if there are no other digits dialed, then the user has completed dialing.
Creation of a local PSTN gateway dial plan is easy (and mostly painless).
Gateways that are based on Skinny Gateway Protocol and MGCP will have their
dial plan information configured within Cisco CallManager itself, whereas H.323
gateways will require only a small set of dial peers.The dial peers are then used
by the gateway to direct calls destined for the PSTN through the Cisco
CallManager.
If you are located outside North America, the numbers of digits that must be
dialed for call completion differ. In this case, you will need to create multiple
length dial-plans.The problem is, with the current version of Cisco CallManager,
the system doesn’t know when the dialing is complete, so you need to create
specific route patterns.
The Session Target
The Session Target
The session target is the IP address of the router to which the call will be
directed once the dial peer is matched. In a VoIP network, you need to configure
this using the session target command under the destination-pattern configuration.
For dial peers that are outbound, the destination-pattern is the telephone
number associated with the device you want to connect to.On inbound dial
peers, the session target is ignored.
The session target is the IP address of the router to which the call will be
directed once the dial peer is matched. In a VoIP network, you need to configure
this using the session target command under the destination-pattern configuration.
For dial peers that are outbound, the destination-pattern is the telephone
number associated with the device you want to connect to.On inbound dial
peers, the session target is ignored.
Character Representations Character Description
Character Representations
Character Description
. This character represents a single digit. Ex 707….
(where …. equals four following digits).
[] These characters represent a range of digits. If the – is used
such as [4–7] then the digits will be consecutive. If a comma is
used, like in [4,7], then the range is nonconsecutive. You can
also use a combination of each [4–7,9].
Note: this only works for single digits [4–7] not [37–41].
() These characters represent a pattern, 425(707). They are
normally used with the ?, %, and/or the +.
? This character is used to specify that the previous digit
happened zero or one time(s) (to use this character you must
use the Ctrl+v key combination).
% This character is used to specify that the previous digit
happened zero or one time(s). It acts like an asterisk (*) and is
used in a regular expression.
+ This character specifies that the previous digit occurred one or
more times.
T This character specifies the timeout used by the interdigit
command.
* or # These characters are standard on touch-tone telephones and
can be used within the dial pattern or as a signal that the user
is done dialing digits using the dial-peer terminator command.
$ This character, when used at the end of a dial string, will
disable variable-length matching for the dial pattern.
Character Description
. This character represents a single digit. Ex 707….
(where …. equals four following digits).
[] These characters represent a range of digits. If the – is used
such as [4–7] then the digits will be consecutive. If a comma is
used, like in [4,7], then the range is nonconsecutive. You can
also use a combination of each [4–7,9].
Note: this only works for single digits [4–7] not [37–41].
() These characters represent a pattern, 425(707). They are
normally used with the ?, %, and/or the +.
? This character is used to specify that the previous digit
happened zero or one time(s) (to use this character you must
use the Ctrl+v key combination).
% This character is used to specify that the previous digit
happened zero or one time(s). It acts like an asterisk (*) and is
used in a regular expression.
+ This character specifies that the previous digit occurred one or
more times.
T This character specifies the timeout used by the interdigit
command.
* or # These characters are standard on touch-tone telephones and
can be used within the dial pattern or as a signal that the user
is done dialing digits using the dial-peer terminator command.
$ This character, when used at the end of a dial string, will
disable variable-length matching for the dial pattern.
Route Pattern (On-Net)
Route Pattern (On-Net)
If you are working with multiple sites across a Wide Area Network with connections
like frame or dedicated circuits, you have the ability to implement on-net
Calls. On-net calls are when you make a call that remains within the network
infrastructure.When using on-net, you have the ability to use abbreviated dialing
string in order to complete calls to other offices.This is just for ease of dialing to
the end user. As an example, let’s say you have an office in Seattle that has a
number range of (206) 707-0000 through (206) 707-0999.You would only need
a single route pattern of 70XXX to complete a call to the Seattle office.The
benefit of this is that it only requires one route pattern entry since the Xs work
as wildcards.
The Cisco CallManager will use route patterns to add or remove digits to the
dialed number.The reason for this is that all dialed strings filtered though the
CallManager must have the appropriate number of digits in order to reach
remote sites (even those located on the same WAN).The Cisco CallManager
simply routes the calls based on these addresses.This is also done to make sure
incoming call numbers don’t need to be changed.
If the WAN cannot complete calls (either due to no connectivity or lack of
sufficient bandwidth), the call will be routed over the PSTN (yet another reason
for the route patterns). In some instances, you will need to have an area code
added to the dial-string.When Cisco CallManager was first released, it was only
able to prepend one set of numbers to any dialed string. Because of this, you had
to use the Cisco IOS gateway to insert the area code (and in some instances, the
three-digit exchange). Cisco fixed that with the release of Cisco CallManager
3.0, which can now add or remove numbers based on a per-route-group basis.
So, you can now manage the entire system from one centralized point that can
control the Cisco IOS gateways (and gateways that use the Skinny Gateway
protocol as well).
www.syngress.com
If you are working with multiple sites across a Wide Area Network with connections
like frame or dedicated circuits, you have the ability to implement on-net
Calls. On-net calls are when you make a call that remains within the network
infrastructure.When using on-net, you have the ability to use abbreviated dialing
string in order to complete calls to other offices.This is just for ease of dialing to
the end user. As an example, let’s say you have an office in Seattle that has a
number range of (206) 707-0000 through (206) 707-0999.You would only need
a single route pattern of 70XXX to complete a call to the Seattle office.The
benefit of this is that it only requires one route pattern entry since the Xs work
as wildcards.
The Cisco CallManager will use route patterns to add or remove digits to the
dialed number.The reason for this is that all dialed strings filtered though the
CallManager must have the appropriate number of digits in order to reach
remote sites (even those located on the same WAN).The Cisco CallManager
simply routes the calls based on these addresses.This is also done to make sure
incoming call numbers don’t need to be changed.
If the WAN cannot complete calls (either due to no connectivity or lack of
sufficient bandwidth), the call will be routed over the PSTN (yet another reason
for the route patterns). In some instances, you will need to have an area code
added to the dial-string.When Cisco CallManager was first released, it was only
able to prepend one set of numbers to any dialed string. Because of this, you had
to use the Cisco IOS gateway to insert the area code (and in some instances, the
three-digit exchange). Cisco fixed that with the release of Cisco CallManager
3.0, which can now add or remove numbers based on a per-route-group basis.
So, you can now manage the entire system from one centralized point that can
control the Cisco IOS gateways (and gateways that use the Skinny Gateway
protocol as well).
www.syngress.com
Usage of the Destination-Pattern
Usage of the Destination-Pattern
To associate a dialed string with a specific telephony device, you would use the
destination-pattern.With it, the dialed string will compare itself to the pattern
and then be routed to the voice port or the session target (discussed later) voice
network dial peer. If the call is an outbound call, the destination-pattern could
also be used to filter the digits that will be forwarded by the router to the
www.syngress.com
AVVID Dial Plans • Chapter 9 291
telephony device or the PSTN.A destination-pattern must be configured for
each and every POTS and VoIP dial peer configured on the router.
You could describe the destination as an entire number or just a partial
number with digits that can be defined through the wildcard switch.The wildcard
digit “.” represents an individual digit the router will be expecting to receive.
If a destination patter is defined as 707…. , then all dialed digits that start with
707 and have four following digits will match this dial peer.
The “.” is not the only character that can be used to represent other digits.
Several others are listed in Table 9.5, along with a brief description, to assist you
in the configuration of your dial peers.
To associate a dialed string with a specific telephony device, you would use the
destination-pattern.With it, the dialed string will compare itself to the pattern
and then be routed to the voice port or the session target (discussed later) voice
network dial peer. If the call is an outbound call, the destination-pattern could
also be used to filter the digits that will be forwarded by the router to the
www.syngress.com
AVVID Dial Plans • Chapter 9 291
telephony device or the PSTN.A destination-pattern must be configured for
each and every POTS and VoIP dial peer configured on the router.
You could describe the destination as an entire number or just a partial
number with digits that can be defined through the wildcard switch.The wildcard
digit “.” represents an individual digit the router will be expecting to receive.
If a destination patter is defined as 707…. , then all dialed digits that start with
707 and have four following digits will match this dial peer.
The “.” is not the only character that can be used to represent other digits.
Several others are listed in Table 9.5, along with a brief description, to assist you
in the configuration of your dial peers.
Dial Peers for Inbound and Outbound Calls
Dial Peers for Inbound and Outbound Calls
Inbound and outbound calls use dial peers to receive and complete calls.You
must remember that the definition of inbound and outbound is based on the
perspective of the router.What this means is that a call coming into the router is
considered an inbound call while a call originating from the router is considered
an outbound call.
When an inbound call is destined for a device on the packet network and is
coming from a POTS interface, the router will match the dial peers for the voice
network with the inbound call leg so it is properly routed to the outbound port.
If the call originates within the packet network, then the router will match the
POTS dial peer and a voice network dial peer so it can modify its attributes for
VAD, CODEC, and QoS.
Routers that receive inbound POTS calls are destined for outbound voice
network dial peers, it will forward all of the collected digits. For outbound POTS
calls, the router will remove explicitly matched digits and forward the remaining
digits to the destination port.
The following configuration is a basic example of POTS and VoIP peers:
dial-peer voice 1 pots
destination-pattern 707....
port 1/0:1
dial-peer voice 2 voip
destination-pattern 707....
session target ipv4:10.1.100.1
As you can see, the router will choose a dial peer for a call leg by matching
the digits defined by the destination-pattern, but it can also use the answeraddress
or incoming called-number commands if they are used within the dial
peer configuration. Be aware that the character “.” is the only wildcard applied if
you use answer-address or incoming call-number commands for the creation of
your dial peers.
Inbound and outbound calls use dial peers to receive and complete calls.You
must remember that the definition of inbound and outbound is based on the
perspective of the router.What this means is that a call coming into the router is
considered an inbound call while a call originating from the router is considered
an outbound call.
When an inbound call is destined for a device on the packet network and is
coming from a POTS interface, the router will match the dial peers for the voice
network with the inbound call leg so it is properly routed to the outbound port.
If the call originates within the packet network, then the router will match the
POTS dial peer and a voice network dial peer so it can modify its attributes for
VAD, CODEC, and QoS.
Routers that receive inbound POTS calls are destined for outbound voice
network dial peers, it will forward all of the collected digits. For outbound POTS
calls, the router will remove explicitly matched digits and forward the remaining
digits to the destination port.
The following configuration is a basic example of POTS and VoIP peers:
dial-peer voice 1 pots
destination-pattern 707....
port 1/0:1
dial-peer voice 2 voip
destination-pattern 707....
session target ipv4:10.1.100.1
As you can see, the router will choose a dial peer for a call leg by matching
the digits defined by the destination-pattern, but it can also use the answeraddress
or incoming called-number commands if they are used within the dial
peer configuration. Be aware that the character “.” is the only wildcard applied if
you use answer-address or incoming call-number commands for the creation of
your dial peers.
Optional Commands for the Configuration of VoIP Command Description
Optional Commands for the Configuration of VoIP
Command Description
Gatekeeper(config-dial-peer)# (Optional) This command chooses the
answer-address string inbound dial peer based on the callingnumber.
Gatekeeper(config-dial-peer)# (Optional) This command chooses the
incoming called-number string inbound dial peer based on the callednumber,
to identify voice and modem calls.
Gatekeeper(config-dial-peer)# (Optional) This command is used to
dtmf-relay [cisco-rtp] configure the tone that sounds in response
[h245-signal] to a pressed digit on a touch-tone
[h245-alphanumeric] telephone.
Dual Tone Multi-Frequency (DTMF) tones are
compressed at one end of a call and
decompressed at the other.
Be aware that if you use a low-bandwidth
CODEC, such as G.729 or G.723, the tones
can sound distorted, which may lead to
problems. The dtmf-relay command transports
DTMF tones generated after call
establishment out-of-band. It uses a
method that sends with greater reliability
than what is possible in-band for most lowbandwidth
CODECs.
Without DTMF Relay, calls established with
low-bandwidth CODECs may experience
trouble accessing automated telephone
menu systems such as voice mail and
Interactive Voice Response (IVR) systems.
A signaling method is supplied only if the
remote end supports it. Options are the
Cisco proprietary Real Time Protocol
(cisco-rtp), standard H.323 (h245-
alphanumeric), and H.323 standard with
signal duration (h245-signal).
Gatekeeper(config-dial-peer)# (Optional) This command indicated the
fax rate {2400 | 4800 | 7200 | transmission speed of a fax to be sent to
9600 | 12000 | 14400 | this dial peer. The keyword disable turns
disable | voice} off fax transmission capability. The keyword
voice, which is on by default, specifies the
highest possible transmission speed
supported by the voice rate.
www.syngress.com
Table 9.4 Optional Commands for the Configuration of VoIP
Command Description
Continued
AVVID Dial Plans • Chapter 9 289
Gatekeeper(config-dial-peer)# (Optional) This command indicates the
numbering-type {abbreviated | numbering type to match, as defined by
international | national | the ITU Q.931 specification.
network | reserved |
subscriber | unknown}
Gatekeeper(config-dial-peer)# (Optional) This command indicates the type
playout-delay mode of jitter buffer playout delay to use.
{adaptive | fixed}
Gatekeeper(config-dial-peer)# (Optional) This command indicates the
playout-delay {maximum amount of time a packet will be held in the
value | nominal value | jitter buffer before it is played out on the
minimum {default | low | audio path.
high}}
Gatekeeper(config-dial-peer)# (Optional) This command configures the
preference value preference for the VoIP dial peer.
The value is a number from 0 through 10.
The lower the number, the higher the
preference.
Gatekeeper(config-dial-peer)# (Optional) This command indicates a
tech-prefix number particular technology prefix that will be
prepended to the destination-pattern of
this dial peer.
Gatekeeper(config-dial-peer)# (Optional) This command indicates the
translate-outgoing {called | translation rule set that needs to be
calling} name-tag applied to the calling-number or
called-number.
Gatekeeper(config-dial-peer)# (Optional) This command enables voice
vad activity detection (VAD). This will disable
the transmission of packets during periods
of silence. VAD is on by default.
The minimum time of silence detection for
VAD can be configured by using the voice
vad-time global configuration command.
The music threshold can be configured by
using the music-threshold voice-port
command, if you feel it is affecting VAD
performance.
Command Description
Gatekeeper(config-dial-peer)# (Optional) This command chooses the
answer-address string inbound dial peer based on the callingnumber.
Gatekeeper(config-dial-peer)# (Optional) This command chooses the
incoming called-number string inbound dial peer based on the callednumber,
to identify voice and modem calls.
Gatekeeper(config-dial-peer)# (Optional) This command is used to
dtmf-relay [cisco-rtp] configure the tone that sounds in response
[h245-signal] to a pressed digit on a touch-tone
[h245-alphanumeric] telephone.
Dual Tone Multi-Frequency (DTMF) tones are
compressed at one end of a call and
decompressed at the other.
Be aware that if you use a low-bandwidth
CODEC, such as G.729 or G.723, the tones
can sound distorted, which may lead to
problems. The dtmf-relay command transports
DTMF tones generated after call
establishment out-of-band. It uses a
method that sends with greater reliability
than what is possible in-band for most lowbandwidth
CODECs.
Without DTMF Relay, calls established with
low-bandwidth CODECs may experience
trouble accessing automated telephone
menu systems such as voice mail and
Interactive Voice Response (IVR) systems.
A signaling method is supplied only if the
remote end supports it. Options are the
Cisco proprietary Real Time Protocol
(cisco-rtp), standard H.323 (h245-
alphanumeric), and H.323 standard with
signal duration (h245-signal).
Gatekeeper(config-dial-peer)# (Optional) This command indicated the
fax rate {2400 | 4800 | 7200 | transmission speed of a fax to be sent to
9600 | 12000 | 14400 | this dial peer. The keyword disable turns
disable | voice} off fax transmission capability. The keyword
voice, which is on by default, specifies the
highest possible transmission speed
supported by the voice rate.
www.syngress.com
Table 9.4 Optional Commands for the Configuration of VoIP
Command Description
Continued
AVVID Dial Plans • Chapter 9 289
Gatekeeper(config-dial-peer)# (Optional) This command indicates the
numbering-type {abbreviated | numbering type to match, as defined by
international | national | the ITU Q.931 specification.
network | reserved |
subscriber | unknown}
Gatekeeper(config-dial-peer)# (Optional) This command indicates the type
playout-delay mode of jitter buffer playout delay to use.
{adaptive | fixed}
Gatekeeper(config-dial-peer)# (Optional) This command indicates the
playout-delay {maximum amount of time a packet will be held in the
value | nominal value | jitter buffer before it is played out on the
minimum {default | low | audio path.
high}}
Gatekeeper(config-dial-peer)# (Optional) This command configures the
preference value preference for the VoIP dial peer.
The value is a number from 0 through 10.
The lower the number, the higher the
preference.
Gatekeeper(config-dial-peer)# (Optional) This command indicates a
tech-prefix number particular technology prefix that will be
prepended to the destination-pattern of
this dial peer.
Gatekeeper(config-dial-peer)# (Optional) This command indicates the
translate-outgoing {called | translation rule set that needs to be
calling} name-tag applied to the calling-number or
called-number.
Gatekeeper(config-dial-peer)# (Optional) This command enables voice
vad activity detection (VAD). This will disable
the transmission of packets during periods
of silence. VAD is on by default.
The minimum time of silence detection for
VAD can be configured by using the voice
vad-time global configuration command.
The music threshold can be configured by
using the music-threshold voice-port
command, if you feel it is affecting VAD
performance.
Options for the Configuration of Dial Plans for VoIP Dial Peers
Options for the Configuration of Dial Plans for VoIP Dial Peers
There are also some configurable options to help you shape the deployment of
your dial peers.Table 9.4 is a list of some of the most common customization
commands.
There are also some configurable options to help you shape the deployment of
your dial peers.Table 9.4 is a list of some of the most common customization
commands.
Dial Peer Commands for Implementing VoIP Command Description
Dial Peer Commands for Implementing VoIP
Command Description
Gatekeeper(config-dial-peer)# This command enters dial-peer
dial-peer voice number voip configuration mode and will define a
remote VoIP dial peer.
The argument number is one or more digits
used to identify the dial peer. Valid entries
are from 1 to 2147483647.
The keyword voip specifies a dial peer that
uses voice encapsulation on the IP network.
Gatekeeper(config-dial-peer)# This command configures the dial peer’s
destination-pattern string [T] destination-pattern so the system can
resolve dialed digits with a telephone
number.
The argument string is a series of digits
used to specify the addressing is E.164, or
the private dialing plan telephone number.
Valid entries are the numbers 0 through 9
and the letters A through D.
Gatekeeper(config-dial-peer)# This command is used to define the IP
session target {ipv4: address of the router, which is connected
destination-address | to the remote telephony device.
dns:[$s$. | $d$. | $e$.|
$u$.] host-name}
www.syngress.com
Table 9.2 Continued
Continued
AVVID Dial Plans • Chapter 9 287
The keyword and argument ipv4:
destination-address indicate the IP address
of the remote router.
The keyword and argument dns:host-name
indicates that the domain name server will
resolve the name of the IP address. Valid
entries for this parameter are characters
representing the name of the host device.
Wildcards are also available for defining
domain names with the keyword by using
source, destination, and dialed information
in the host name.
Gatekeeper(config-dial-peer)# This command defines the CODEC for the
codec {g711alaw | g711ulaw | dial peer.
g723ar53 | g723ar63 | The optional switch bytes will set the
g723r53 | g723r63 | g726r16 | number of voice data bytes per frame.
g726r24 | g726r32 | g728 | Values are from 10 to 240 in increments
g729br8 | g729r8 [pre-ietf]} of 10 (for example, 10, 20, 30, and so on)
[ bytes] are considered acceptable. Any other value
is rounded down (for example, from 144 to
140).
The CODEC value must be matched on
both VoIP dial peers on either side of the
connection.
If you specify g729r8, then IETF bit-ordering
will be used.
Be aware that the CODEC command syntax
is platform- and release-specific.
Command Description
Gatekeeper(config-dial-peer)# This command enters dial-peer
dial-peer voice number voip configuration mode and will define a
remote VoIP dial peer.
The argument number is one or more digits
used to identify the dial peer. Valid entries
are from 1 to 2147483647.
The keyword voip specifies a dial peer that
uses voice encapsulation on the IP network.
Gatekeeper(config-dial-peer)# This command configures the dial peer’s
destination-pattern string [T] destination-pattern so the system can
resolve dialed digits with a telephone
number.
The argument string is a series of digits
used to specify the addressing is E.164, or
the private dialing plan telephone number.
Valid entries are the numbers 0 through 9
and the letters A through D.
Gatekeeper(config-dial-peer)# This command is used to define the IP
session target {ipv4: address of the router, which is connected
destination-address | to the remote telephony device.
dns:[$s$. | $d$. | $e$.|
$u$.] host-name}
www.syngress.com
Table 9.2 Continued
Continued
AVVID Dial Plans • Chapter 9 287
The keyword and argument ipv4:
destination-address indicate the IP address
of the remote router.
The keyword and argument dns:host-name
indicates that the domain name server will
resolve the name of the IP address. Valid
entries for this parameter are characters
representing the name of the host device.
Wildcards are also available for defining
domain names with the keyword by using
source, destination, and dialed information
in the host name.
Gatekeeper(config-dial-peer)# This command defines the CODEC for the
codec {g711alaw | g711ulaw | dial peer.
g723ar53 | g723ar63 | The optional switch bytes will set the
g723r53 | g723r63 | g726r16 | number of voice data bytes per frame.
g726r24 | g726r32 | g728 | Values are from 10 to 240 in increments
g729br8 | g729r8 [pre-ietf]} of 10 (for example, 10, 20, 30, and so on)
[ bytes] are considered acceptable. Any other value
is rounded down (for example, from 144 to
140).
The CODEC value must be matched on
both VoIP dial peers on either side of the
connection.
If you specify g729r8, then IETF bit-ordering
will be used.
Be aware that the CODEC command syntax
is platform- and release-specific.
Configuring Dial Peers for VoIP
Configuring Dial Peers for VoIP
By configuring VoIP dial peers, you can enable the router to make outbound calls
to other telephony devices, located within the network. In order to configure a
dial peer for VoIP, you need to assign a unique tag to the dial peer for identification,
define the destination telephone number, and define the destination IP
address.Table 9.3 shows some basics.
By configuring VoIP dial peers, you can enable the router to make outbound calls
to other telephony devices, located within the network. In order to configure a
dial peer for VoIP, you need to assign a unique tag to the dial peer for identification,
define the destination telephone number, and define the destination IP
address.Table 9.3 shows some basics.
Optional Dial Plan Configuration Commands Command Description
Optional Dial Plan Configuration Commands
Command Description
Gatekeeper(config-dial-peer)# (Optional) This command selects the
answer-address string inbound dial peer based on the callingnumber.
Gatekeeper(config-dial-peer)# (Optional) This command selects the
incoming called-number string inbound dial peer based on the callednumber
so it can identify voice and
modem calls.
Gatekeeper(config-dial-peer)# (Optional) This command enables Direct
direct-inward-dial string Inward Dialing (DID) call treatment for the
incoming called-numbers.
Gatekeeper(config-dial-peer)# (Optional) This command configures digitforward-
digits { num-digit | forwarding for the dial peer. The valid
all | extra} range for the number of digits that can be
forwarded (num-digit) is 0 through 32.
Gatekeeper(config-dial-peer)# (Optional) This command specifies the
max-conn number maximum number of connections allowed
to and from the POTS dial peer. The valid
range is 1 through 2147483647.
Gatekeeper(config-dial-peer)# (Optional) This command specifies which
numbering-type {abbreviated | numbering type to match, as defined by
international | national | the ITU Q.931 specification.
network | reserved |
subscriber | unknown}
Gatekeeper(config-dial-peer)# (Optional) This command configures the
preference value preference for the POTS dial peer.
The valid range is 0 through 10. The lower
the number, the higher the preference is
for that dial peer.
Gatekeeper(config-dial-peer)# (Optional) This command adds a prefix that
prefix string the system will prepend to the dial string
before passing it to the telephony interfaces.
Valid entries for the string argument are 0
through 9 and a comma. You would use
the comma to include a one-second pause
between digits to allow for a secondary
dial tone.
www.syngress.com
Continued
286 Chapter 9 • AVVID Dial Plans
Command Description
Gatekeeper(config-dial-peer)# (Optional) This command specifies the
translate-outgoing {called | translation rule set that is applied to the
calling} name-tag calling-number or called-number.
Command Description
Gatekeeper(config-dial-peer)# (Optional) This command selects the
answer-address string inbound dial peer based on the callingnumber.
Gatekeeper(config-dial-peer)# (Optional) This command selects the
incoming called-number string inbound dial peer based on the callednumber
so it can identify voice and
modem calls.
Gatekeeper(config-dial-peer)# (Optional) This command enables Direct
direct-inward-dial string Inward Dialing (DID) call treatment for the
incoming called-numbers.
Gatekeeper(config-dial-peer)# (Optional) This command configures digitforward-
digits { num-digit | forwarding for the dial peer. The valid
all | extra} range for the number of digits that can be
forwarded (num-digit) is 0 through 32.
Gatekeeper(config-dial-peer)# (Optional) This command specifies the
max-conn number maximum number of connections allowed
to and from the POTS dial peer. The valid
range is 1 through 2147483647.
Gatekeeper(config-dial-peer)# (Optional) This command specifies which
numbering-type {abbreviated | numbering type to match, as defined by
international | national | the ITU Q.931 specification.
network | reserved |
subscriber | unknown}
Gatekeeper(config-dial-peer)# (Optional) This command configures the
preference value preference for the POTS dial peer.
The valid range is 0 through 10. The lower
the number, the higher the preference is
for that dial peer.
Gatekeeper(config-dial-peer)# (Optional) This command adds a prefix that
prefix string the system will prepend to the dial string
before passing it to the telephony interfaces.
Valid entries for the string argument are 0
through 9 and a comma. You would use
the comma to include a one-second pause
between digits to allow for a secondary
dial tone.
www.syngress.com
Continued
286 Chapter 9 • AVVID Dial Plans
Command Description
Gatekeeper(config-dial-peer)# (Optional) This command specifies the
translate-outgoing {called | translation rule set that is applied to the
calling} name-tag calling-number or called-number.
Options for the Configuration of Dial Plans for POTS Dial Peers
Options for the Configuration of Dial Plans for POTS Dial Peers
There are also some configurable options to help you shape the deployment of
your dial peers.Table 9.2 is a list of some of the most common customization
commands.
There are also some configurable options to help you shape the deployment of
your dial peers.Table 9.2 is a list of some of the most common customization
commands.
Basic Configuration Commands for the Creation of Dial Peers Command Description
Basic Configuration Commands for the Creation of Dial Peers
Command Description
Gatekeeper(config-dial-peer)# This allows you to enter dial-peer
voice number pots configuration mode so you can define a
local dial peer connecting to a POTS interface.
The argument number is one or more
digits that can identify the dial peer. Valid
entries are from 1 to 2147483647.
The keyword pots specifies that the dial
peer is using basic telephone service.
Gatekeeper(config-dial-peer)# This command matches dialed digits to a
destination-pattern string [T] telephony device.
The argument string is a series of digits
that specify the addressing is E.164, or a
private dialing plan telephone number.
Valid entries are the numbers 0 through 9
and the letters A through D.
When the timer (T) character is included
at the end of the destination-pattern, the
router then collects dialed digits until the
interdigit timer expires, approximately ten
seconds if left at the default, or until you
dial the termination character. Usually this
is left as the default.
Be aware the timer character must be a
capital T for it to work.
Gatekeeper(config-dial-peer)# This command maps a dial peer to a
port location specific logical interface that it needs to be
associated with. Be aware that the port
command syntax is platform-specific.
Command Description
Gatekeeper(config-dial-peer)# This allows you to enter dial-peer
voice number pots configuration mode so you can define a
local dial peer connecting to a POTS interface.
The argument number is one or more
digits that can identify the dial peer. Valid
entries are from 1 to 2147483647.
The keyword pots specifies that the dial
peer is using basic telephone service.
Gatekeeper(config-dial-peer)# This command matches dialed digits to a
destination-pattern string [T] telephony device.
The argument string is a series of digits
that specify the addressing is E.164, or a
private dialing plan telephone number.
Valid entries are the numbers 0 through 9
and the letters A through D.
When the timer (T) character is included
at the end of the destination-pattern, the
router then collects dialed digits until the
interdigit timer expires, approximately ten
seconds if left at the default, or until you
dial the termination character. Usually this
is left as the default.
Be aware the timer character must be a
capital T for it to work.
Gatekeeper(config-dial-peer)# This command maps a dial peer to a
port location specific logical interface that it needs to be
associated with. Be aware that the port
command syntax is platform-specific.
Configuring Dial Peers for POTS
Configuring Dial Peers for POTS
When you configure a dial peer for POTS, you need to assign a unique tag to
the dial peer for identification, define the destination (either a telephone number
www.syngress.com
284 Chapter 9 • AVVID Dial Plans
or a range of telephone numbers), and then associate a voice port so calls can be
established.Table 9.1 shows the standard configurations.
When you configure a dial peer for POTS, you need to assign a unique tag to
the dial peer for identification, define the destination (either a telephone number
www.syngress.com
284 Chapter 9 • AVVID Dial Plans
or a range of telephone numbers), and then associate a voice port so calls can be
established.Table 9.1 shows the standard configurations.
Configuring Dial Peers for Use
Configuring Dial Peers for Use
Configuring dial peers is essential when designing and implementing Voice over
IP on your network. Dial peers identify the calling source and the destination
points so as to define what attributes are assigned to each call. In the telecommunications
world, calls routed over the PSTN are assigned to a dedicated 64KB
circuit from start to end. In the data world, a voice call must traverse segments
within the network referred to as call legs.This isn’t to say that these don’t exist
in the telecomm world, they are just more noticeable within packet-based
networks.
A call leg is the connection that occurs between the calling device and the
router, router-to-router along the path, and from the router to the end device.A
dial peer is linked with each of these segments, and it’s here where the defined
attributes are added to the call.Things like the CODEC,VAD, and QoS are utilized
depending on whether you have defined them for that link.
There are several types of dial peers, but those we’ll focus on here are VoIP
and POTS.VoIP dial peers are associated with the IP address of the destinations
router so it can connect and terminate a call from an IP-based telephony device.
POTS dial peers are basically the telephone system as we know it.These dial
peers map dialed digit strings to specific voice ports that are located on the
router.These voice ports are usually associated with the PSTN and the PBX.
Configuring dial peers is essential when designing and implementing Voice over
IP on your network. Dial peers identify the calling source and the destination
points so as to define what attributes are assigned to each call. In the telecommunications
world, calls routed over the PSTN are assigned to a dedicated 64KB
circuit from start to end. In the data world, a voice call must traverse segments
within the network referred to as call legs.This isn’t to say that these don’t exist
in the telecomm world, they are just more noticeable within packet-based
networks.
A call leg is the connection that occurs between the calling device and the
router, router-to-router along the path, and from the router to the end device.A
dial peer is linked with each of these segments, and it’s here where the defined
attributes are added to the call.Things like the CODEC,VAD, and QoS are utilized
depending on whether you have defined them for that link.
There are several types of dial peers, but those we’ll focus on here are VoIP
and POTS.VoIP dial peers are associated with the IP address of the destinations
router so it can connect and terminate a call from an IP-based telephony device.
POTS dial peers are basically the telephone system as we know it.These dial
peers map dialed digit strings to specific voice ports that are located on the
router.These voice ports are usually associated with the PSTN and the PBX.
Subscribe to:
Posts (Atom)