Session Manager Case Study Book

Avaya Aura
Session Manager Case

Studies
Issue 3
03-603478
Release 6.0
June 2010
2010 Avaya Inc.

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2 Session Manager Case Studies June 2010
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Session Manager Case Studies June 2010 3
Contents
Chapter 1: Overview..................................................................................................................7
Chapter 2: A Network Case Study...........................................................................................9
Overview...........................................................................................................................................................9
The network......................................................................................................................................................9
Core provisioning.............................................................................................................................................11
Domains..................................................................................................................................................11
SIP entities for Session Managers.........................................................................................................12
SIP entity for Westminster Session Manager.........................................................................................12
SIP entity for NJ Session Manager.........................................................................................................13
Locations.........................................................................................................................................................15
Locations with managed bandwidth.......................................................................................................15
Locations without managed bandwidth..................................................................................................16
Time ranges....................................................................................................................................................17
Non-Session Manager SIP entities.................................................................................................................17
Harmonizing disparate PBXs..........................................................................................................................18
Adaptations for PBXs.............................................................................................................................19
SIP entities for PBXs..............................................................................................................................24
SIP service providers......................................................................................................................................30
SIP service provider adaptations............................................................................................................30
AT&T adaptation.....................................................................................................................................30
Verizon adaptation..................................................................................................................................31
Hypothetical adaptation..........................................................................................................................32
SIP entities for SIP service providers..............................................................................................................33
Single SIP entity behind SBC.................................................................................................................34
Multiple SIP entities behind SBC............................................................................................................35
Routing policies for SIP service providers.......................................................................................................36
Simple routing policy for routing policies for SIP service providers........................................................37
Alternative routing policy for routing policies for SIP service providers..................................................38
Dial patterns for SIP service providers............................................................................................................39
Simple routing policy for dial patterns for SIP service providers example..............................................40
Alternative routing policy for dial patterns for SIP service providers......................................................41
Tail-end hop-off................................................................................................................................................42
SIP foundation servers....................................................................................................................................45
Modular Messaging................................................................................................................................45
Voice Portal-like SIP application service................................................................................................48
Chapter 3: A New User Setup Case Study............................................................................53
Overview.........................................................................................................................................................53
Synchronize Communication Manager station data to the System Manager.................................................53
Adding a SIP entity for the Session Manager.................................................................................................56
Adding the Session Manager Instance...........................................................................................................57
Adding Domains..............................................................................................................................................58
Adding Application Sequences.......................................................................................................................58
Adding a Home Location.................................................................................................................................58
Adding the Survivability Server.......................................................................................................................59
Adding a User (SIP end-point)........................................................................................................................60
General section......................................................................................................................................61
Identity section........................................................................................................................................61
Address section......................................................................................................................................62
Communication Profile section...............................................................................................................62
Session Manager section.......................................................................................................................64
Endpoint Profile section (CM Station association)..................................................................................65
For a new station....................................................................................................................................66
Default Contact List................................................................................................................................66
Chapter 4: A Call Handling Case Study.................................................................................67
Overview.........................................................................................................................................................67
Scenario Definition..........................................................................................................................................67
Using Application Sequence...........................................................................................................................69
Adding an Application (e.g. CM Feature Server)............................................................................................69
Creating an Application Sequence from Existing Applications........................................................................71
Administering Implicit Users............................................................................................................................72
Index.........................................................................................................................................75
Chapter 1: Overview
This document explains different scenarios about the functionality of Avaya Aura
Session
Manager which provide a practical understanding on administering and configuring Session
Manager. This should be read in conjunction with the book Administering Avaya Aura
Session Manager, 03-603324. The book covers the following case studies:
1. A Network Case Study
2. A New User Setup Case Study
3. A Call Handling Case Study
Refer to the book Administering Avaya Aura
Communication Manager Server Options,

03-603479 for understanding different Communication Manager Server roles in the Session
Manager environment.
Intended Audience
This book is meant for all those who are involved in either understanding, administering and
troubleshooting Session Manager.
Overview
Chapter 2: A Network Case Study
Overview
The following case study describes a network which uses Session Manager to provide the
following solutions:
Harmonizing disparate PBXs, both extension lengths and brands
VoIP connections to SIP service providers (access to the public switched telephone
network [PSTN] via SIP signaling)
Tail-end hop-off - maintaining calls on the internal core network as long as possible,
hopping off to the PSTN at a point where calls are local and/or where they possibly cost
less.
Access to SIP foundation servers including Avaya Aura
Modular Messaging.
These solutions make use of the following Session Manager features:
Geographically redundant network session control structure
Least cost routing
Alternate routing around network faults based on active SIP monitoring
Network bandwidth use limitation based on session admission control
Load balancing
Session (or call) detail recording.
The network
This Case Study network consists of:
Two Session Managers in the core network for redundancy
Communication Managers in Westminster, Highlands Ranch, New Jersey HQ, and Avaya
Labs New Jersey with differing length (3, 4, and 5-digit) dial plans
Cisco CallManager in San Jose with a 5-digit dial plan
A separate SIP trunk to the AT&T SIP service provider
A session border controller through which trunks to Verizon and hypothetical SIP service
providers can be accessed
Modular Messaging system that serves all users in the enterprise
A Voice Portal Service for 1-866-GO-Avaya provided in separate locations in the network.
Before beginning to administer Session Manager, we must decide:
Domains that are used for routing. We make Session Manager authoritative for both
avaya.com and avayalabs.com. The avayalabs.com domain is used for calls originated
from the Avaya Labs NJ Communication Manager.
The enterprise-wide dial plan and any domain-specific dial plans. Each user on one of
the PBXs can dial another user, local or remote, through a unique 7-digit enterprise-
canonical number.
The locations that are defined for call admission control and any location-specific dial
plans.
We adopt a basic philosophy to guide us in administering adaptation and the dial plan:
The Session Manager dial plan routes internal enterprise-wide numbers and E.164
numbers (including E.164 representations of internal numbers).
Calling party numbers are sent from the local PBXs in their local dial plan format. These
numbers are all converted to enterprise-canonical on ingress to the Session Manager.
Called party numbers are sent from the local PBXs in their local dial plan format. These
numbers are all converted to either enterprise-canonical numbers or E.164 on ingress to
the Session Manager.
Calling party numbers that are in enterprise-canonical format are converted to whatever
the service provider requires when a request is forwarded by the Session Manager.
After some initial, core provisioning, for each solution, we follow the administration of this
configuration in the order recommended in Routing. This suggests defining in order, domains,
locations, adaptations, SIP entities, SIP entity links, time ranges, routing policies, and dialing
patterns.
A Network Case Study
Core provisioning
Core provisioning includes:
The domains for which Session Manager is authoritative. However, these can be added
as more are created.
The SIP entities for the Session Manager servers. You can add these as more entities
are created, linking other SIP entities as appropriate.
Locations used to group entities for differing dial plans and/or bandwidth management.
Again, you can add these as necessary.
Related topics:
Domains on page 11
SIP entities for Session Managers on page 12
SIP entity for Westminster Session Manager on page 12
SIP entity for NJ Session Manager on page 13
Domains
First, we add the two domains that appear in the request-URI of INVITE messages sent by the
Communication Managers:
The Cisco PBX and the service providers send the IP address of the Session Manager in the
request-URI. Later, we administer the Session Manager to convert this IP address into the
avaya.com domain so that calls from these entities can be routed.
Core provisioning
SIP entities for Session Managers
Though routing suggests finishing the locations and adaptations before beginning SIP entities,
Session Manager typed SIP entities are special in that they are not associated with a location
nor an adaptation. The other SIP entities normally have both associations. Another, possibly
more natural, course is to provision the location, adaptation and SIP entity detail for each SIP
entity in turn rather than doing all locations and adaptations before proceeding to the SIP
entities.
SIP entity for Westminster Session Manager
Adaptation and location are not necessary for Session Manager instances which essentially
define the core network. These are necessary only for other SIP entities.
Generally, Session Manager listens for connections on ports specified in the entity link table,
which is detailed later in this case study. If a port is specified in the Session Manager's sip
entity port table (as shown below), SIP messages which contain this Session Manager's IP
address in their Request-URI have it replaced by the domain specified in the port table. The
Cisco PBX and some service providers can send a request with the Session Manager's IP
address in the Request-URI. This port table entry is also currently necessary if SIP monitoring is
to be used to monitor connectivity between Session Manager instances within the core
network.
SIP entity for NJ Session Manager
This differs only in minor ways from the Westminster Core Session Manager. A network with
two Session Manager instances would normally have each SIP entity connecting to both
instances so that one instance can act as backup for the other in case of network or Session
Manager failure.
Core provisioning
So that sessions can be alternately routed through another Session Manager in the case of
network failure of connectivity between a non-Session Manager SIP entity and a particular
Session Manager, the Session Manager instances need to be connected. This connection, like
all inter-SIP entity connections, is specified with an entity link entry.
Currently, all entity links should be marked as Trusted or else they will not function. Unless
there are restrictions against its use, the inter Session Manager entity link should use the
Protocol TLS. If TLS is used by other SIP entities then it must be used, the reason being that
SIP security standards do not allow passing secure information (like media encryption keys)
over a connection where one of the inter-proxy connections is not TLS. Using TLS works,
however, even if a SIP entity does not support TLS on its connection to Session Manager.
If there were three Session Manager instances in this case study, there would need to be an
entity link between each pair, for a total of three.
Locations
Putting SIP entities into different locations currently serves two purposes:
Provides a way to use different dialing plans. Calls originating from SIP entities in different
locations can match different dialing pattern entries and route differently even though the
dialed address are precisely the same.
Can be used to limit the bandwidth used between the core network and that location.
In this case study each edge SIP entity is given its own location. Locations can be created and
the location with which a given SIP entity is associated can be changed at any time.
Related topics:
Locations with managed bandwidth on page 15
Locations without managed bandwidth on page 16
Locations with managed bandwidth
In this case study the Westminster location exemplifies one where bandwidth is managed by
setting the Managed Bandwidth: value to only allow 100 simultaneous calls in or out of
Westminster. The current release of Session Manager assumes that each call to and from the
location use the Average Bandwidth per Call amount of network bandwidth. Note that the
calls that stay within the location, even if they route through Session Manager, are not counted.
Thus, the number of simultaneous calls allowed to and from that location are calculated by
dividing the Managed Bandwidth value by the Average Bandwidth per Call value. The two
values may be scaled differently, so this might not be a simple division. In the example below,
the Managed Bandwidth is 8000 Kbits/s and the Average Bandwidth per Call is 80 Kbits/
s, so the calculation is simple (8000/80 => 100). But if the Managed Bandwidth was instead 8
Mbits/s, then the 8 would need to be scaled to 8000 Kbits/s before the calculation is performed.
Incoming calls are associated with SIP entities in varying ways. If the call is associated with a
particular SIP entity, it is deemed to have come from the location associated with that SIP entity.
If a location contains location pattern IP address patterns, it can override the location
association with the SIP entity. If a SIP entity's IP address is listed explicitly in the IP address
patterns, as in this example, then there is no doubt about with which location it is associated.
Locations
Locations without managed bandwidth
The Research location exemplifies one without managed bandwidth. This does not mean there
is unlimited bandwidth between this location and the core, just that Session Manager does not
manage the bandwidth and does not limit the number of calls it sends to this location. There
may be other limiting factors (like the number of calls the SIP entity will accept) to which Session
Manager reacts. Simply leaving the Managed Bandwidth: field blank keeps Session Manager
from managing the bandwidth.
The advantage to having Session Manager manage bandwidth is that it recognizes bandwidth
exhaustion to a particular location before attempting to route a call there and it can then perform
alternate routing earlier than if it had to wait for the SIP entity at that end to tell it that bandwidth
was not available. Additionally Session Manager can associate multiple SIP entities with a
given location and manage the bandwidth to the entire location where each SIP entity would
not know the full bandwidth status of the location.
Time ranges
These are used for alternate routing. They are not associated with any particular SIP entity,
but are needed for the specification of the ranking of routing preferences (even if routing does
not depend upon the time of day). It is useful to have defined at least the All Day time range.
Non-Session Manager SIP entities
The provisioning details of the non-Session Manager SIP entities is outlined as the solutions
with which they are associated are explained below. The resulting SIP entities in this case
study are listed here in the SIP Entities table. Note that currently the Type of the entity matters
only if it is Session Manager. All other types are effectively the same, though this could change
in future releases, so it is best to choose an appropriate type.
Time ranges
In this case study non-Session Manager SIP entities fall into one of three categories:
PBXs that front telephones and PSTN trunks
SIP service providers that essentially front PSTN trunks. Even though they might route
some calls entirely within their SIP network it is assumed all or a known subset of PSTN
numbers can be reached through them.
SIP foundation servers such as Avaya Modular Messaging or Voice Portal or any other
server that creates communication sessions with SIP.
Harmonizing disparate PBXs
This case study has two brands of PBXs: Communication Manager and Cisco Call Manager.
Additionally the PBXs have different length extensions (or dial plans). These two aspects
require the Session Manager to adapt (that is, alter) SIP messages to the standard SIP
messaging performed by Session Manager as well as the E.164 and Enterprise Canonical
(that is, common) dial plan used in this example.
Related topics:
Adaptations for PBXs on page 19
SIP entities for PBXs on page 24
Adaptations for PBXs
Adaptation is normally needed for all of the PBX devices that connect to the Session Manager.
They, like locations, can be created before a given PBXs SIP entity. This is particularly useful if
two PBXs might use the same adaptation. Alternatively, one can create SIP entities with blank
adaptations first and then modify the SIP entity to use a particular adaptation created later.
Adaptation digit conversion is likely the most complex aspect of and requires the most planning
for the deployment of this example.
Related topics:
Westminster PBX adaptation on page 19
NJ HQ Communication Manager adaptation on page 20
Avaya Labs research PBX adaptation on page 21
San Jose PBX adaptation on page 22
Westminster PBX adaptation
The Westminster Communication Manager has a local 5-digit dial plan (8xxxx). Each extension
can also be dialed from other systems using the 7-digit enterprise canonical number 538-
xxxx. The PBX also has DID numbers assigned; a PSTN caller can dial +1303538xxxx to reach
an extension.
This adaptation uses the DigitConversionAdapter. The Westminster PBX is set up to be
authoritative for dr.avaya.com on its network region form. This means that INVITEs sent from
Session Manager to the PBX must have dr.avaya.com in the host part of the request-URI. The
odstd parameter to the adaptation module specifies this. The PBX also uses dr.avaya.com as
the far-end domain on the signaling group to the Session Manager, which means that the P-
Asserted-Identity header of incoming INVITEs must be changed to dr.avaya.com. We use the
osrcd parameter to the adaptation module to accomplish this.
The digit conversion tables are set up accordingly. The text that is placed in the Adaptation
Module box is DigitConversionAdapter odstd=dr.avaya.com osrcd=dr.avaya.com (the
parameter osrcd means override source domain).
NJ HQ Communication Manager adaptation
The NJ HQ Communication Manager has a local 4-digit dial plan (xxxx). Each extension can
also be dialed from other systems using the 7 digit enterprise canonical number 953-xxxx. The
PBX also has DID numbers assigned; a PSTN caller can dial +1908953xxxx to reach an
extension.
This adaptation uses the DigitConversionAdapter. The NJ HQ PBX is set up to be authoritative
for nj.avaya.com on its network region form. It also uses nj.avaya.com as the far-end domain
on the signaling group to the Session Manager. These domain conversions are specified as
parameters to the adaptation module. The text that is placed in the Adaptation Module box is
DigitConversionAdapter odstd=dr.avaya.com osrcd=dr.avaya.com
Avaya Labs research PBX adaptation
The Avaya Labs Communication Manager has a local 3-digit dial plan (xxx). Each extension
can also be dialed from other systems using the 7-digit enterprise canonical number 696-5xxx.
The PBX also has DID numbers assigned; a PSTN caller can dial +19086965xxx to reach an
extension.
This adaptation uses the DigitConversionAdapter. The Communication Manager is set up to
be authoritative for avayalabs.com on its network region form. It also uses avayalabs.com as
the far-end domain on the signaling group to the Session Manager, which means that the P-
Asserted-Identity header of incoming INVITEs must be changed to avayalabs.com.
The text that is placed in the Adaptation Module box is DigitConversionAdapter
odstd=avayalabs.com osrcd=avayalabs.com
San Jose PBX adaptation
The San Jose PBX has a local 5-digit dial plan (1xxxx). This means that extensions connected
to this PBX normally dial each other by entering only five digits. Each extension can also be
dialed from other systems using the 7-digit enterprise-canonical number 661-xxxx. The PBX
also has DID (direct inward dialing) numbers assigned; a PSTN caller can dial +1408661xxxx to
reach an extension. For calls made by local users, adaptation converts
Numbers dialed by local users into enterprise canonical numbers (for example, a San
Jose user calls another San Jose user via the Session Manager)
Local Calling party numbers into enterprise-canonical numbers (for example, San Jose
user calls a Westminster user. The calling party number displayed in Westminster is the
enterprise-canonical number)
Calls to international numbers to E.164 format (for example, someone dials 011+digits)
Calls to North American numbers to E.164 format
For calls made to the San Jose PBX, adaptation must convert:
The called party enterprise-canonical number into a local extension number (for example,
a 661-xxxx number into a 1xxxx number)
The E.164 number into a local extension number, for calls coming from a service provider
(for example, +1408661xxxx into 1xxxx)
This adaptation uses the CiscoAdapter to convert between the proprietary headers Cisco uses
to convey display and diversion information with the standard headers used by Avaya products.
CiscoAdapter, like all currently available adapters, can also perform digit conversion. The Cisco
PBX also needs its IP address (135.9.106.161) as the host part of the request-URI, so this
conversion is specified as a parameter (odstd, which means override destination domain).
The digit conversion tables convert between the local 1xxxx dial plan and the enterprise
canonical 661-xxxx dial plan. On ingress to the Session Manager, digit strings in the local 1xxxx
dial plan need to be converted into the enterprise canonical form. On egress from the Session
Manager, the 661-xxxx enterprise canonical numbers need to be converted into the local
extensions.
The digit conversion tables also convert dialed numbers for international calls (011+digits) or
calls within North America (10 digit, 1+10 digit) to E.164 form when requests enter the Session
Manager. Similarly, E.164 and enterprise canonical calls to local users must be converted into
local form on egress from the Session Manager.
SIP entities for PBXs
The PBXs in this case study are shown in the SIP entity table above as type Communication
Manager or Other (though not all Other-typed SIP entities are edge PBXs). They are each
administered similarly. The FQDN/IP Address is the primary distinguishing attribute. It is used
to identify sessions as originating from that SIP entity and to where it should send messages
to establish sessions to the entity. The location associated with the SIP entity (unless
overridden) factors into the routing and the adaptation affects how the messages may be
modified.
Related topics:
Single interface on page 24
Multiple interfaces on page 25
Routing policies for PBXs on page 27
Dial patterns for PBXs enterprise canonical numbering on page 29
Single interface
The Westminster Communication Manager exemplifies one with a single interface, where its
IP address is specified. The other PBXs: Basking Ridge HQ, Basking Ridge Research, and
San Jose (even being a Cisco Call Manager) are all similar.
Generally, PBXs do not need call detail records created for each call, so the Call Detail
Recording: field is set to none. SIP monitoring should be used for all SIP entities that support it
(and most all PBXs do), and so the value of Monitoring on/off: should be left as Use Session
Manager configuration, which allows specification of global (to this Session Manager instance)
SIP monitoring parameters.
To define the ports and transport types supported by this SIP entity,Entity Links entries are
made. The entity link table for this SIP entity shows that this SIP entity connects to both Session
Manager instances in the network.
In both cases TCP port 5060 is used, not only for connections out to the SIP entity, but
connections in from it. SIP Entity 1 is always a Session Manager SIP entity. Conventionally,
the Name, which must be unique for all links in the system, contains some representation of
the name of the Session Manager SIP entity and that of the non-Session Manager SIP entity,
though this is not necessary. Currently, the entity link must always be marked as Trusted or
else it fails to function.
Multiple interfaces
The Highlands Ranch Communication Manager shows a SIP entity with multiple interfaces.
This might be typical of a large Communication Manager with more than one C-LAN. It allows
for redundant routing and a higher level of fault tolerance. Routing to it is handled by specifying
an FQDN for it, which resolves to multiple IP addresses.
This particular SIP entity supports TLS, so the entity link protocol and port are adjusted
accordingly.
The FQDN in this case can either be resolved by DNS or through a locally provisioned FQDN to
IP address mapping. The latter mapping is specified in the Session Manager Element
manager's Local Host Name Resolution table
This table is used both for simple FQDN to IP address mapping (like this example) plus full
port and transport specification (shown later). For the simple case, the Port and Transport
values are ignored because they are specified in the entity link table (5061 and TLS). The
convention of giving the Port a value of 1 is used to indicate this. The Priority is used, and
given that the first line has a higher priority (lower numbered value), it is always used first. The
only time the second entry is used is if a failure is encountered while using the first IP address.
To realize this connectivity, the HighlandsRanch Communication Manager requires four
signaling and trunk groups: one from each C-LAN to each of the Session Managers. The first
signaling group:
The transport method and port match those specified in the entity link from each Session
Manager to the Communication Manager. The IMS Enabled field should not be set for standard
SIP scenarios. Level 3 tests enabled has the Communication Manager perform OPTIONS
monitoring of the Session Manager instances. The Bypass request should probably not be
enabled. Communication Manager tests if the network characteristics between its media
processors and the Session Manager are suitable for media and Communication Manager
should not be sending media to the Session Manager. The four signaling groups:
Assuming all the SIP trunk groups associated with the signaling groups used the same group
number, an outgoing routing pattern would look like:
The Communication Manager would try each of its connections to its local Session Manager
(PBX and Session Manager NA:US:CO are both in CO) before trying the remote Session
Manager (NA:US:NJ in NJ). The LAR (look-ahead routing) field must be next on every
preference that needs to skip to the next route on an error.
Routing policies for PBXs
Routing policies indicate the rank order of a particular SIP entity. Multiple routing policies can be
associated with a dial pattern (as shown later) to specify alternate routing. Additionally, the
rank of a routing policy can be changed by the time of day. For simple PBX routing (like in this
example) specific dial patterns are associated with only one PBX and these do not vary by the
time of day. For these types of routing policies the convention is to use the SIP entity's name
as the routing policy name too.
The routing policy detail allows the association of the routing policy with the SIP entity. There
is no time-of-day routing associated with this policy so only the All Day time range is added to
the Time of Day section. Also this is not to be part of an alternate route so Ranking of 0 is used.
The dial patterns are shown here, but they are defined on another form.
Dial patterns for PBXs enterprise canonical numbering
The dial patterns defined in this solution allow the on-net 7-digit dialing between the disparate
PBXs.
In the details for the dial pattern, a blank Domain indicates the pattern matches any domain.
The originating location and routing policy associated with this dial pattern are -ALL- and the
Westminster PBX SIP entity respectively. This means a session created from anywhere with
a user part of 538xxxx routes to this particular PBX. Specific originating locations could be
denied access to this dial pattern (that is, the call would be denied).
SIP service providers
This case study depicts three SIP service providers: AT&T, Verizon, and Hypothetical. AT&T
is connected through one session boarder controller (though it could be connected directly)
while Verizon and Hypothetical are connected through a different, though shared SBC.
For connections to SIP service providers to be useful for outgoing as well as incoming traffic,
each PBX must, in addition to routing their on-net 7-digit calls, route their PSTN calls to the
Session Manager core. As seen before the 7-digit calls are routed to other PBXs while the
PSTN numbers are converted to E.164 and routed to SIP service providers (as seen in this
section).
Related topics:
SIP service provider adaptations on page 30
AT&T adaptation on page 30
Verizon adaptation on page 31
Hypothetical adaptation on page 32
SIP service provider adaptations
SIP service providers typically send in digit strings formatted for the calling area in which the
service is provided. In North America they may be 10- or 7-digit called and calling party
numbers. For outgoing calls (relative to the Session Manager network) they may allow only
calling party numbers from a block of purchased ones (that is, those they route into the Session
Manager network over the SIP facility). Adaptation can be used for the necessary conversions.
AT&T adaptation
When a request comes in from AT&T, the calling and called party numbers is 10 digits for calls
originated from North America and E.164 for international calls. Session Manager converts the
called party number to E.164 form.
When sending requests to AT&T, Session Manager has to convert calling and called party
numbers. AT&T requires that the host part of the request-URI be their IP address (in this
example, 135.9.43.69). The called party number must be sent in the request-URI as either
011+ digits for international calls or as a 10-digit North American number. The calling party
number must be sent as a 10-digit number.
This adaptation uses the AttAdapter, which does digit conversion and strips the History-Info
header from requests, as the AT&T network is not compatible with this header that is used by
Communication Manager. The IP address supplied by AT&T must be specified as a parameter
to convert the host-part of the request-URI. Thus, the text to enter in the Adaptation Module
box is AttAdapter odstd=135.9.43.69.
Verizon adaptation
Verizon adaptation is similar to AT&T adaptation, with these differences:
Verizon uses the VerizonAdaptation adaptation module. This module does digit
conversion and converts the History-Info header to a Diversion header and vice-versa.
In this case study the Verizon connection and the following Hypothetical service provider
connection is done through a common session border controller (SBC). The conversion
of the domain in the SIP message request URI to the IP address of the Verizon gateway is
handled by the SBC.
SIP service providers
Hypothetical adaptation
The basic DigitConversionAdapter may suffice for adapting connectivity to other SIP service
providers. Alternatively, some adaptation functions could be provided by an external SBC
which is used in this particular case study, leaving the DigitConversionAdapter to simply
convert digit fields in the SIP message.
When a request comes in from Hypothetical, the calling and called party numbers are 10 digits
for calls originated from North America and E.164 for international calls. Session Manager
converts the called party number to the E.164 form.
When sending requests to Hypothetical, Session Manager has to convert calling and called
party numbers. Hypothetical requires that the host part of the request-URI be hyposp.com
and that DNS be used to locate the correct server. The called party number must be sent in
the request-URI as either 011+ digits for international calls or as a 1+10 digit North American
number. The calling party number must be sent as a 10-digit number.
When calls originate from one of the PBXs, the calling party numbers are in the enterprise-
canonical format. On egress to Hypothetical, these numbers must be converted to 10-digit
North American numbers. The Digit Conversion for Outgoing Calls table below has been set
up to do this. The choice of the origination address in the Address to modify column indicates
that only calling party numbers (in the P-Asserted-Identity header) of the request is modified.
The choice of the destination in the rule that converts E.164 to North American format indicates
that only the Request-URI is modified.
This adaptation uses the DigitConversionAdapter. The hyposp.com domain must be specified
as a parameter to convert the host-part of the request-URI. Also, Hypothetical requires the
'user=phone' parameter in the request-URI. This is entered in the Egress URI Parameters
box on the form.
Note that the Adaptation Module and Egress URI Parameters fields allow free format text.
These fields scroll horizontally so their complete values may not show on the form without
active selection and scrolling (as seen in this particular example). Additionally, the system does
not validate the values in these fields, so enter these values carefully.
SIP entities for SIP service providers
SIP service provider connections differ from edge PBXs in various ways. Normally they are
connected through session border controllers, and they require call detail recording. In this
case study there are three SIP service provider connections: one, AT&T, is connected through a
dedicated SBC while the other two, Hypothetical and Verizon, share an SBC. Other minor
differences are highlighted.
Related topics:
Single SIP entity behind SBC on page 34
Multiple SIP entities behind SBC on page 35
Single SIP entity behind SBC
A single connection behind an SBC has the characteristic that the IP address of the SBC can be
considered to represent the one and only SIP entity behind it. There is no reason to distinguish
messages coming from the SBC from those intended for the SIP entity. The SIP entity
configuration looks similar to that of any other SIP entity.
In this example the IP address of the SBC is used. The Type is marked SIP Trunk for
categorization.
The Call Detail Recording: field is set at egress, meaning that all sessions established out to
this SIP entity are recorded. Incoming sessions are not recorded, unless they route out to a
SIP entity that is marked for egress or both CDR recording.
Again, an entity link must be created for each Session Manager to this SIP entity, in this case
UDP is the Protocol used, though the SBC could be used to translate the UDP supported by
AT&T to TCP. The SBC must also be willing to accept messages from each Session Manager
and route messages to either Session Manager (in case of network or Session Manager
failure).
Multiple SIP entities behind SBC
In this example, both the Verizon and Hypothetical service providers are behind an SBC at an
IP address specified by an FQDN. We do not have an entry in the Local Host Name Resolution
table for this. The Session Manager goes to the network's DNS server to resolve the IP
address.
The entity links specified for this SIP entity use the standard ports for TCP.
The Hypothetical service provider's gateway, behind the SBC, is located in the same location as
that of Verizon, so any bandwidth management for that location applies to both SIP entities.
The time zone, used for time-of-day routing differs. Although this may sound strange (same
location, but different time zone), it is an artifact of grouping them together for bandwidth
management. Both, in this case study, are reached over the same physical communication link
to the SBC, but behind that, their respective gateways are located in different time zones and
the tariffs they specify have different rates depending upon the time zone in which the gateways
are located. Additionally, Hypothetical does not support any form of SIP monitoring, so this is
disabled on the SIP entity form.
The special entity link administration is what makes this configuration possible. Notice that the
ports are different and nonstandard on both sides of the entity link. The SBC must be
programmed to send messages from Hypothetical's gateway to port 5062 using the TLS
protocol and must be willing to accept connections to port 5062 and forward those messages to
Hypothetical (the ports could have been different). Using the different ports is what allows the
Session Manager to distinguish the traffic from the same SBC IP address to be from different
SIP entities.
Routing policies for SIP service providers
The routing polices Alternate-AT&T, Alternate-Verizon, and HypotheticalSP are the primary
policies that route to the SIP service providers. The Alternate-BaskingRidge polices are for
tail-end hop-off (discussed later), and the Ap800 polices are for some SIP foundation servers
(also discussed later).
Related topics:
Simple routing policy for routing policies for SIP service providers on page 37
Alternative routing policy for routing policies for SIP service providers on page 38
Simple routing policy for routing policies for SIP service providers
The HypotheticalSP is a simple routing policy. No alternate routing to it is anticipated, so it
has a single Rank 0 time of day entry. A unique dial pattern references this policy and no other.
Routing policies for SIP service providers
Alternative routing policy for routing policies for SIP service
providers
The Alternate-AT&T and Alternate-Verizon routing policies are meant to work in concert. The
convention used here is to name such routing policies with the Alternate prefix and put their
relative ranking in the Notes field. Here the Alternate-AT&T policy has a 10 ranking while the
Alternate-Verizon has 20, so any dial pattern that references both of these routing policies
prefers Alternate-AT&T.
Another convention is to rank policies in increments of 10 at the start. This allows insertion of
intermediately ranked policies without having to renumber all that would come later.
Note that when routing sessions, the Session Manager chooses the lower ranked routing policy
for one of three reasons:
SIP monitoring has declared all endpoints identified by the all higher ranked policies as
down.
This call fails to route (due to a bad return code or TimerB failure).
This call fails to route due to managed bandwidth being exhausted to the destination
location.
Dial patterns for SIP service providers
The E.164 dial pattern entries exist mainly because our case study network is connected to
the PSTN through SIP service providers. They fall into four categories, marked in the Notes
field:
APP route directly to a SIP foundation server (see below). These are marked.
TEHO for Tail-End Hop-Off (see below)
E.164 route DID numbers from the SIP service provider to the proper PBX. The relatively
low number of entries of this type result from the PBXs in question owning the full bank
of DID numbers (for example, the Westminster PBX owns all numbers +13035380000 to
+13035389999). If this were not the case, the discrepancy could be solved with more
entries, either with a larger set of more explicit ones to route fewer numbers to the given
PBX, or with more explicit ones that route exceptions elsewhere (like the +13035389077
entry).
PSTN (the + entries) are very general patterns which essentially match any E.164 number
not matched by a more explicit entry.
As shown in all the adaptations, all the PBXs and incoming service provider calls have their
destination addresses converted to E.164 (that is, a + is prepended), so that matching dial
patterns can be more uniform and predictable.
Note that the first + entry in the table below is distinct from the second, because it only applies if
the destination has a domain of avayalabs.com, while the second pattern matches any other
domain.
The four entries:
Related topics:
Simple routing policy for dial patterns for SIP service providers example on page 40
Alternative routing policy for dial patterns for SIP service providers on page 41
Simple routing policy for dial patterns for SIP service providers
example
The simple case dictates that any session destined for any E.164 address with the domain
avayalabs.com routes only using the HypotheticalSP routing policy. In this case study the
avayalabs.com domain is used by the BaskingRidge:Research SIP entity. Therefore all E.
164 numbers that do not match any of the other patterns are routed using the HypotheticalSP
SIP service provider.
Alternative routing policy for dial patterns for SIP service providers
This dial pattern entry performs two distinct route selections for all E.164 numbers (except
those to the avayalabs.com domain). If the originating location is SanJose, then the
HypotheticalSP service provider is used. Any other originating location selects (using alternate
routing) the Alternate-AT&T and Alternate-Verizon routing policies.
Tail-end hop-off
TEHO builds on the E.164 routing to the SIP service providers and relies on the same
fundamental assumption that PBXs route their PSTN calls into the Session Manager core.
Select E.164 patterns route first or exclusively to a PBX which has PSTN trunks of its own to
handle the call. In this case study both BaskingRidge PBXs can handle calls to the 908 area
code. The dial pattern entry identifies one of four route policies.
The Alternate-AT&T and Alternate-Verizon entries were shown before. They have ranks for 10
and 20, respectively. The other entries Alternate-BaskingRidge:Research and Alternate-
BaskingRidge:HQ are shown below to have time-of-day varying ranks.
BaskingRidge:Research has a rank of 1 on weekdays and a rank of 2 on weekends and
BaskingRidge:HQ has a rank of 2 on weekdays and a rank of 1 on weekends.
Tail-end hop-off
Thus the routing policy (and so SIP entity) ordering is as such:
Weekdays:
- BaskingRidge:Research
- BaskingRidge:HQ
- AT&T
- Verizon
Weekends:
- BaskingRidge:HQ
- BaskingRidge:Research
- AT&T
- Verizon
It is desirable for the two BaskingRidge PBXs to be able to make use of the SIP service provider
trunks even for 908 area code calls if their PSTN trunks are in use or out of service. To do this,
and to make configuration of the PBXs routing simpler they route all their PSTN calls into the
Session Manager core. A potential problem is when the 908 area code calls get routed back
to them. If no consideration is made for this, the calls are simply routed back to the Session
Manager core only to potentially loop back into the very same PBX. To avoid this, adaptation
for the PBXs changes the destination address so that the PBX can recognize it as needing
routing out of its PSTN trunk rather than back into the Session Manager core.
Adaptation for BaskingRidge:Research:
The +1908 entry modifies all 908 area code numbers to be destined for *9-000-1-908-xxx-
xxxx. The PBX must recognize this to be a specially routed number. In the case of this
Communication Manager, the *9 is the ARS access code and the 000 is an otherwise
nondialable digit string that can be used for this unique identification.
SIP foundation servers
The two SIP application types shown in this case study are Modular Messaging and Voice
Portal. Both of these applications handle incoming and outgoing calls, although Modular
Messaging primarily handles incoming calls. Like any SIP entity, adaptation for digit conversion
can be used, but that is needed primarily for existing Modular Messaging or Voice Portal
applications with existing dial plans. New applications can be provisioned to use the full length
E.164 numbers (save possibly to delete and add the +) internally.
Related topics:
Modular Messaging on page 45
Voice Portal-like SIP application service on page 48
Modular Messaging
On each Communication Manager system, Modular Messaging is associated with a hunt group
and assigned a routing digit string to route covered and direct sessions with media and a Voice
Mail handle for message waiting indication subscriptions and notifications. Both types of
sessions must be routed by Session Manager from each supported Communication Manager
to the appropriate Modular Messaging system.
On the Communication Managers in this case study the following hunt group data is entered:
The number 3035389077 is routed through a dial pattern entry:
that identifies a SIP entity. CDR can be on, and it might be interesting to track calls into the
Modular Messaging system, though the Modular Messaging system itself has an internal CDR
capability. It may also be useful to implement bandwidth management on the Modular
Messaging system in its own location, but it has also has a way of limiting calls. What is
interesting is the load balancing between the multiple message access servers (MASs) within a
Modular Messaging system. And in this particular case the local host name resolution table is
used to provide the multiple IP addresses as well as the port and transport information needed
to contact the MASs.
With entity links from both the Session Managers, checking Override Port & Transport with
DNS SRV on the SIP entity form indicates that both the Port and Protocol (aka Transport) on
the SIP entity form are ignored. The convention used here is that a Port value of 1 indicates
that both are ignored.
SRV records within the DNS server accessed by the Session Managers could be used to
provide the necessary overridden information, but it is much easier to include this information in
the Local Host Name Resolution table:
In this particular case TLS connections are made to port 5061 of both of the MASs at the
indicated IP addresses. Load balancing is done on a statistically weighted basis, because each
MAS is at the same priority. About 10/30 (or one third) of the calls are given to the MAS at
135.9.43.34 while two-thirds of the calls go to the other. This ratio is valid over many calls.
There is a possibility that for any given small set of calls, more or less than 1/3 of the calls go to
the first MAS.
The message waiting indication subscribe and notify sessions are routed with the handle
specified in Communication Manager ([email protected] in this case). Handles are currently
routed using the regular expressions table:
The pattern here is very precise with no meta character pattern matching symbols only because
a specific handle needs to be matched. The fewer the meta characters, the more efficient the
match. The routing policy selected by this match is the same one selected by the dial pattern of
the number associated with the Communication Manager hunt group (that is, +13035389077).
Voice Portal-like SIP application service
The other SIP application service shown in this case study is similar to how calls are routed
to a Voice Portal. The Voice Portal administration itself is not shown nor is this application fully
set up to make outgoing calls. Adaptation would most likely be necessary for this.
This Voice Portal-like application is reached when 1-866-GO-AVAYA is dialed. The dial pattern
is quite complex:
To paraphrase the routing policy selection, without actually listing the routing policies
themselves:
SIP entities in the locations Tokyo and Sydney prefer the APAC:App800 foundation server
(a SIP entity) and falls back to the NA:US:App800 server. There are, however, no SIP
entities associated with these locations yet.
All other SIP entities (including the ones defined in this case study PBX and SIP service
provider alike) prefer the NA:US:App800 server.
SIP entities in the SanJose, Westminster, and BaskingRidge:Research locations cannot
route calls to this dial pattern. The calls are denied.
The SIP entities for the foundation servers are similar.
They have different FQDNs, are in different locations and time zones, but they both choose to
override the port and transport specified in the entity link.
The Local Host Name Resolution table shows why this override is necessary at least in the
case of the NA:US:App800 SIP entity. The APAC:App800 SIP entity just has one associated
IP address that uses the standard TLS port.
The NA:US:App800 SIPentity chooses one of four different server IP addresses based on a
total weight of 100. Ten percent of the time it goes to 135.9.95.101 with TCP, 30% to
135.9.95.100 with TLS, 50% to 135.9.43.29 with TLS, and the remaining 10% to 135.9.95.102
with UDP.
Chapter 3: A New User Setup Case Study
Overview
Fred Flintstone just joined our Highlands Ranch office. He is the new Network Engineer and
will be working in the Network Management group. The following case study describes the
necessary steps for adding Fred as a user and registering his user profile with a Session
Manager for accessing enhanced enterprise call handling facilities through
an application sequence (with CM feature server and other applications)
modular messaging mailbox
telephone set
thereby providing the option of choosing his preferred communication devices.
Before user setup is done, it is important to synchronize the CM Station Data to System
Manager as shown in the section Synchronize CM station data to the System Manager. The
various stages for the user setup are listed as follows:
1. Adding a SIP entity for the Session Manager (with listen ports)
2. Adding the Session Manager instances (for both primary and secondary Session
Manager)
3. Adding SIP Domains
4. Adding Application Sequences
5. Adding Survivability Server
6. Adding Home Location
7. Adding a User (SIP end-point)
Synchronize Communication Manager station data to the
System Manager
The Communication System Manager interface is used to synchronize Communication
Manager station data to the System Manager database.
1. Each Communication Manager must be administered as an Entity using the
Manage Elements web page located at Elements > Inventory > Manage Elements
in the navigation menu. The main page shows all of the administered entities and
for those corresponding to Communication Manager, the Type column indicates
CM.
2. Click the New button and select CM to add a new Communication Manager entity. In
the New CM Instance page, under the Application section, enter the name of the
Communication Manager instance and specify the management IP address for the
Communication Manager (the address that is used for SAT login) in the Node field.
3. The default (none) is OK for the SNMP Attributes section.
A New User Setup Case Study
4. Finally, the Communication Manager SAT login information must be configured in
the Attributes section. For most users (for SSH SAT login), enter the SAT login in
the Login field and the associated password in the Password field. Click the
Commit button on the bottom of the page.
5. Automatic CM Data Synchronization After a Communication Manager has been
added as an entity, it is automatically scheduled for an initial and subsequent
incremental daily data synchronization. The synchronization status can be viewed
for each Communication Manager using Elements > Inventory > Synchronization
> Communication System in the navigation menu. Each Communication Manager
is displayed in the table as shown below. The Last Sync Time column indicates the
status of when the last data sync completed or the phase of synchronization for the
current sync job in progress.
Synchronize Communication Manager station data to the System Manager
Adding a SIP entity for the Session Manager
1. You need to add a SIP entity as a Session Manager instance and hence you need to
ensure that the SIP entity has been added in the Routing application as a SIP entity
with type Session Manager.
2. If not, you can create a SIP entity using Routing > SIP Entities.
3. You also need to specify Listen Port under Entity Links in Personal Settings screen.
Adding the Session Manager Instance
You need to administer the Session Manager (where Freds phone will be registered)
using Elements > Session Manager > Session Manager Administration.
Adding the Session Manager Instance
Adding Domains
Freds phone number is captured as a SIP handle as [email protected] where part
of the SIP handle is a domain (avaya.com). Therefore, administer the domain using
the Routing application (Routing > Domains). Enter the domain's name and select
type sip
Adding Application Sequences
Fred can have an Origination Application Sequence and a Termination Application Sequence
as his preferred way of handling calls. When a call is made from the user, a Session Manager
will route the call through the origination sequence. Whereas when a call is made to the user, a
Session Manager will route the call through the termination sequence. For details on
configuring an Application Sequence, refer to the Call Handling Case Study.
Adding a Home Location
When this user calls numbers that are not associated with an administered user, dial-plan rules
will be applied to complete the call based on this home location regardless of the physical
location of the SIP device used to make the call.
Note:
Dial plan rules are applied using Routing > Dial Patterns.
Add a location using Routing > Locations as a Home Location element for Fred's
Communication Profile. A Home Location can be specified to support mobility for the
currently displayed user. A selection is mandatory.
Adding the Survivability Server
Prerequisites
To use a Branch Session Manager as a Survivability Server, you need to add a SIP entity of
type Session Manager .
For local survivability, a Survivability Server can be specified to provide survivability
communication services for devices associated with a Communication Profile in the event that
local connectivity to Session Manager instances in the Aura Core is lost. This is optional and
is required only for survivability.
Adding the Survivability Server
For administering Branch Session Manager as a Survivability Server, go to Elements >
Session Manager > Session Manager Administration and click New on the Branch
Session Manager Instances section of Session Manager Administration screen.
Adding a User (SIP end-point)
For basic setup of Fred's user profile, you need to complete the following sections in User
Management application. Navigate to Users > Manage Users , click New and enter the
following information:
1. General
2. Identity
3. Communication Profile
4. Default Contact List
General section
In User Management module, the general section allows you to specify the basic information
about Fred's profile. In the General section,
1. Enter the Fred's last name and first name.
2. Enter a description in the Description field. This field is optional.
Identity section
1. Scroll to the Identity section.
2. Enter a Login Name for Fred. This is the unique system login name given to Fred
and takes the form of username@domain (enterprise canonical number) which is
used to create the Fred's primary handle.
3. Select the Authentication Type as Basic.
4. Enter an System Manager Login Password and confirm it. The password must start
with an alpha (lower or upper case) character.
5. Enter the Shared Communication Profile Password. The password must be in
numeric characters. This is the password that is used when logging in to the phone.
6. Enter the Localized Display Name of the Fred. This is the name that is displayed as
the calling party.
7. Enter the full text name of the user for Endpoint Display Name.
Address section
In the Address section, add mailing address details of Fred in the Identity section.
Communication Profile section
Fred may have one or more Communication Profiles for registering one or more SIP user
handles (phone extensions) to the Session Manager. This will enable Fred to define (optional)
an origination and termination application sequence as his preferred call routing method.
To register a SIP phone with Session Manager, at least one Communication Profile must be
administered containing the Session Manager related details and must have defined at least
one Communication Address of type SIP. The handles may also be associated with a
Communication Manager station and/or messaging subscriber. You can specify the following
information in Fred's Communication Profile:
A communication address can be used to communicate with the contact. This can be a phone
number, e-mail address, SIP or IM of the contact. One or more communication addresses for
Fred is defined in relation to handle plus domain in userinfo@domainpart format when routing a
communication interaction to Fred. For Fred, the communication address is set as shown in
the figure below,
In Communication Address section,
1. Select Type which specifies the type of the handle which is set as Avaya SIP. Types
are specified as followed:
2. Enter the phone extension in the Fully Qualified Address field and finally select
the correct domain from the drop-down menu.
Handle is a unique communication address for Fred which is set as 5380000 and
the name of the domain with which the handle is registered is set as avaya.com.
Session Manager section
In the Session Manager Profile section, you can associate Fred with a Primary and
Secondary Session Manager , specify Origination and Termination Application
Sequences, a Survivability Server (e.g. Branch Session Manager), and also specify a
Home Location for this user. Fred's phones will register with the selected Session
Manager. Calls from or to Fred's phone will be routed through the selected origination
or termination applications sequences respectively.
Home Location is a mandatory input field to support mobile users. Locations are
administered using Routing > Locations.
Endpoint Profile section (CM Station association)
In Station Profile section, specify the Communication Manager station association for Fred as
per the following cases:
1. Associate Fred with an existing station
1. Select the previously administered Communication Manager entity in the System
drop-down menu.
2. Check the Use Existing Endpoints check box.
3. Enter (or select when prompted) the extension for the station.
4. Optionally, the template for the station, security code and/or port values for this
station can be changed.
For a new station
2. Add a new station on the Communication Manager for Fred
1. Select the previously administered Communication Manager entity in the System
drop-down menu.
2. Enter (or select when prompted) the extension for the station.
3. Select a phone template for the Fred's phone.
4. Enter (or select when prompted) a value for the Port.
5. Optionally, enter a value for the Security Code.
Messaging Profile section
In the Messaging Profile section, specify the association of a subscriber mailbox for Fred. You
can include the following details:
1. Add messaging system on which you need to add Fred.
2. Add template (system defined and user defined) you want to associate with Fred.
Templates are defined in the Communication System Management module.
3. Add mailbox number for Fred.
Default Contact List
Fred can optionally be given a default Contact List by expanding the Contact List section toward
the bottom of the page and pressing the Add button to select already administered users as
contacts.
Note:
These contacts are transferred to Freds phone if the phone supports contacts feature.
Chapter 4: A Call Handling Case Study
Overview
This case study shows call handling as per user's preferred set of applications using
Application Sequence functionality or as per some enterprise dial pattern using Implicit User
functionality. Some examples of such applications are as follows:
Do-not call and selective call lists
Caller-ID manipulation Outsourcers or partners calling on behalf of their customer
Conference bridge selector Use built in 6-party conferencing first, then automatically
switch to conference bridge (internal or hosted) if greater capacity needed
Call Screener Could be activated if user is in a meeting or depending on their presence
status
Selective call recording
Call transcription/archiving
Scenario Definition
This scenario shows how a call from Barney for Fred is handled using Fred's Application
Sequencing definition of his Communication Profile at Termination side.
Origination Side Sequencing
1. Barney makes an outbound call.
2. Session Manager signals the originating applications associated with Barney.
3. The DND Application (Do Not Call) verifies that the called party (Fred) is not listed
in Do Not Call registry.
4. After verification, it forwards the call to Session Manager.
5. Session Manager sequences to the next application in the sequence
Termination Side Sequencing
1. Session Manager gets INVITE from Trunk.
2. Session Manager signals the Terminal Applications for the Called Party (Fred).
3. The Call Blocker application does not block call.
A Call Handling Case Study
4. Session Manager signals the next application.
5. The call screening application checks Freds status.
6. If Fred is Busy, call screening application forwards request to Media Server.
7. The Media Server plays custom message accordingly.
Using Application Sequence
Application Sequence functionality enables you to define and manage a set of applications for
call sequencing as per users Communication Profile. Some of the steps are outlined as
follows:
1. You should setup application sequences before users are assigned.
2. All applications should use Communication Manager as part of the sequence.
3. Administer Communication Manager SIP entity beforehand using Routing > SIP
Entities.
4. Associate the user with a particular Session Manager instance and an application
sequence as the originating and terminating sets as shown in the New User Setup
Case Study.
Adding an Application (e.g. CM Feature Server)
1. Add the Feature Server SIP Entity You need to add the Communication Manager
feature server as a SIP Entity in the Routing application.
2. Add the Application The Communication Manager feature server can now be
administered as an application using the Session Manager application.
For administering applications, use Elements > Session Manager >
Application Configuration > Applications. The main page displays a list of
currently administered applications. Click the New button to add a new
Using Application Sequence
application.
Enter a name for the application as ACM and select the associated
Communication Manager feature server for the SIP Entity input. Also select
the CM System for this Communication Manager (you need to add an entity
of type CM previously using Elements > Inventory > Manage Applications )
for the data synchronization to System Manager.
Add other applications to be added in the Fred's Application Sequence.
Note:
By itself, an application cannot be associated with a user. Only application
sequences consisting of one or more applications assigned in an order can be
associated with a user for call routing.
Creating an Application Sequence from Existing
Applications
1. The Application Sequences web page is located below the Applications page
(Elements > Session Manager > Application Configuration > Application
Sequences). The main page displays all currently administered Application
Sequences. Press the New button to add a new sequence.
2. Name the application sequence and click the (+) icon next to an available application
(such as the Communication Manager Feature Server ACM). Now, the selected
application (ACM) will get added in the Applications in this Sequence table. This
Creating an Application Sequence from Existing Applications
suggests that ACM is now a part of the Application Sequence.
Administering Implicit Users
The Implicit User functionality allows routing calls to and/or from a specified dial pattern through
application sequences. This is similar to specifying an origination and termination application
sequence for a user, but it allows application sequencing to be applied to any dial pattern as
opposed to a phone number for a user's registered phone. However at first, a match on an
explicit users dial pattern is attempted. If no match is found, then an attempt is made to match
on an implicit user. Some of the important points about Implicit Users are:
Implicit users are not registered users
Users can be on Third Party PBXs
Also includes DCP, Analog or H.323 CM users
Identified by phone numbers or extensions
Can have Origination and Termination application sequences
To create an Implicit User rule:
1. From the navigation pane on the System Manager Common Console, click
Elements > Session Manager > Application Configuration > Implicit Users to
open the Implicit Users screen.
2. Click New. The Implicit User Rule Editor screen appears. The pattern, min and max
input fields are similar to those on the Dial Pattern web page and are used to specify
a dial pattern. The SIP Domain field allows restricting the origination and termination
application sequencing to calls from or to a matching phone number on the specified
domain only.
3. On the Implicit User Rule Editor screen, enter the appropriate information and click
Commit.
Administering Implicit Users
Index
A
adaptations for PBXs ..................................................19
adding a Home Location ............................................58
Adding a SIP entity for the Session Manager .............56
adding an application .................................................69
Adding Application Sequences ...................................58
Adding Domains .........................................................58
Adding the Session Manager Instance .......................57
Adding the Survivability Server ..................................59
administering Implicit users ........................................72
alternative routing policy for dial patterns for SIP service
providers ........................................................41
alternative routing policy for routing policies for SIP
service providers ...........................................38
Application Sequencing Scenario ...............................68
AT&T adaptation .........................................................30
Avaya Labs research PBX adaptation ........................21
C
core provisioning ........................................................11
creating an Application Sequence ..............................71
D
dial patterns for PBXs enterprise canonical numbering
.......................................................................29
dial patterns for SIP service providers ........................39
domains ......................................................................11
H
harmonizing disparate PBXs ......................................18
hypothetical adaptation ..............................................32
L
legal notice ...................................................................2
locations .....................................................................15
locations with managed bandwidth ............................15
locations without managed bandwidth .......................16
M
Modular Messaging ....................................................45
multiple interfaces ......................................................25
multiple SIP entities behind SBC ................................35
N
network, case study description ...................................9
New User Setup .........................................................53
NJ HQ Communication Manager adaptation ..............20
non-Session Manager SIP entities .............................17
O
overview .......................................................................7
R
Routing
case study ..............................................................9
routing policies for PBXs ............................................27
routing policies for SIP service providers ...................36
S
San Jose PBX adaptation ..........................................22
simple routing policy for dial patterns for SIP service
providers example .........................................40
simple routing policy for routing policies for SIP service
providers ........................................................37
single interface ...........................................................24
single SIP entity behind SBC .....................................34
SIP entities for PBXs ..................................................24
SIP entities for Session Managers .............................12
SIP entities for SIP service providers .........................33
SIP entity for NJ Session Manager ............................13
SIP entity for Westminster Session Manager .............12
SIP foundation servers ...............................................45
SIP service provider adaptations ................................30
SIP service providers .................................................30
Synchronize CM station data to the System Manager . . .
54
T
tail-end hop-off ...........................................................42
time ranges .................................................................17
U
Using Application Sequence ......................................69
V
Verizon adaptation ......................................................31
Voice Portal-like SIP application service ....................48
W
Westminster PBX adaptation .....................................19

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