Avaya Aura Communication Manager, Aura Session Manager SIP trunk solution, 4500 Net-Net Session Director, MetaSphere CFS Application Notes

Avaya Solution & Interoperability Test Lab
Application Notes for Avaya Aura™ Communication
Manager 5.2.1, Avaya Aura™ Session Manager 5.2.1.1, and
Acme Packet 4500 Net-Net Session Director integration with
Metaswitch MetaSphere CFS – Issue 0.1
Abstract
These Application Notes describe the steps to configure an Avaya Aura™ SIP trunk solution
with Metaswitch MetaSphere CFS. The Avaya SIP trunk architecture consists of Avaya
Aura™ Communication Manager (version 5.2.1), and Avaya Aura™ Session Manager
(version 5.2.1.1), and an Acme Packet 4500 Net-Net Session Director (6.1.0).
The Metaswitch MetaSphere CFS solution referenced within these Application Notes is
designed for customers with an Avaya SIP trunk solution. The Metaswitch MetaSphere CFS
solution provides access to service providers for local and/or long Distance PSTN calling via
standards-based SIP trunks directly, without the need for additional TDM enterprise gateways
or TDM cards and the associated maintenance costs.
Information in these Application Notes has been obtained through DevConnect compliance
testing and additional technical discussions. Testing was conducted in the Avaya
Interoperability Test Lab, utilizing a connection to the Metaswitch Test Lab over the public
network.
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Table of Contents
1.
1.1.
1.2.
2.
2.1.
3.
4.
4.1.
4.1.1
4.1.2
4.1.3
4.1.4
4.1.5
4.1.6
4.1.7
4.1.8
4.1.9
4.1.10
4.1.11
5.
5.1.
5.1.1
5.1.2
5.1.3
5.1.4
5.1.5
5.1.6
5.1.7
5.1.8
5.1.9
6.
6.1.
6.2.
6.3.
6.3.1
6.3.2
6.3.3
6.3.4
6.3.5
6.3.6
6.3.7
6.3.8
6.4.
6.5.
Introduction.................................................................................................................4
Interoperability Compliance Testing ..........................................................................4
Support........................................................................................................................5
Reference Configuration.............................................................................................5
Local to Foreign Domain Conversion for Outbound Calls.........................................7
Equipment and Software Validated ............................................................................8
Configure Avaya Aura™ Communication Manager for SIP Trunking......................9
Verify System Capacity and Features.......................................................................10
Dial Plan....................................................................................................................12
Uniform Dialplan ......................................................................................................13
Node Names..............................................................................................................13
IP-Network-Regions .................................................................................................14
IP Codec Sets ............................................................................................................15
SIP Trunk Groups .....................................................................................................16
Public Unknown Numbering – Basic Configuration ................................................21
Call Routing..............................................................................................................22
Avaya Aura™ Communication Manager Stations (non-SIP)...................................25
EC500 Provisioning..................................................................................................27
Save Avaya Aura™ Communication Manager Provisioning...................................27
Configure Avaya Aura™ Communication Manager as a Feature Server for SIP
Trunking....................................................................................................................28
Verify System Capacity and Features.......................................................................28
Dial Plan....................................................................................................................30
Uniform Dialplan ......................................................................................................31
Node Names..............................................................................................................31
IP-Network-Regions .................................................................................................31
IP Codec Sets ............................................................................................................33
SIP Trunk Groups .....................................................................................................34
Private Unknown Numbering – Basic Configuration...............................................39
Call routing ...............................................................................................................40
Save Avaya Aura™ Communication Manager Provisioning...................................41
Avaya Aura™ Session Manager Provisioning .........................................................42
Network Interfaces....................................................................................................42
Logging into System Manager..................................................................................42
Network Routing Policy ...........................................................................................44
SIP Domains .............................................................................................................44
Adaptations ...............................................................................................................45
Locations...................................................................................................................47
SIP Entities................................................................................................................48
Entity Links...............................................................................................................51
Time Ranges .............................................................................................................52
Routing Policies ........................................................................................................53
Dial Patterns..............................................................................................................55
Avaya Aura™ Session Manager...............................................................................57
Feature Server ...........................................................................................................60
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6.6.
7.
7.1.
7.2.
7.3.
7.3.1
7.3.2
7.3.3
7.3.4
7.3.5
7.3.6
7.3.7
7.3.8
7.3.9
7.3.10
7.3.11
7.3.12
8.
8.1.
8.2.
8.3.
8.3.1
8.3.2
8.3.3
9.
10.
10.1.
10.2.
10.2.1
10.2.2
10.3.
10.4.
11.
11.1.
11.2.
11.3.
User Management for Adding SIP Telephone Users................................................65
Acme Packet 3800 Net-Net Session Director...........................................................69
Acme Packet Service States......................................................................................69
Acme Packet Network Interfaces..............................................................................69
Acme Packet Provisioning........................................................................................69
Acme Packet Management .......................................................................................70
Local Policies............................................................................................................71
Network Interfaces....................................................................................................71
Physical Interfaces ....................................................................................................72
Realms.......................................................................................................................73
Steering-Pools ...........................................................................................................73
Session-Agents..........................................................................................................74
Session Groups..........................................................................................................75
SIP Configuration .....................................................................................................75
SIP Interfaces ............................................................................................................76
SIP Manipulation ......................................................................................................76
Other Acme Packet provisioning ..............................................................................78
Metaswitch Configuration ........................................................................................79
Media Gateway Model..............................................................................................79
Configured SIP Binding............................................................................................81
PBX object configuration .........................................................................................82
PBX Object ...............................................................................................................82
PBX Line Object.......................................................................................................83
DID objects ...............................................................................................................83
General Test Approach and Test Results..................................................................84
Verification Steps......................................................................................................84
Verify Avaya Aura™ Communication Manager 5.2................................................85
Verify Avaya Aura™ Session Manager ...................................................................87
Verify SIP Entity Link Status ...................................................................................87
Verify System State ..................................................................................................88
Verification Call Scenarios .......................................................................................89
Conclusion ................................................................................................................90
References.................................................................................................................90
Avaya ........................................................................................................................90
Metaswitch................................................................................................................90
Acme Packet .............................................................................................................90
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1. Introduction
These Application Notes describe the steps to configure an Avaya Aura™ SIP trunk solution with
Metaswitch MetaSphere CFS. The Avaya SIP trunk architecture consists of Avaya Aura™
Communication Manager (version 5.2.1), and Avaya Aura™ Session Manager (version 5.2.1.1),
and an Acme Packet 4500 Net-Net Session Director (6.1.0). Various Avaya analog, digital, H.323,
and SIP stations are also included in the configuration.
The Acme Packet 4500 Net-Net Session Director is used as an edge device between the Avaya
Customer Premises Equipment (CPE) and the Metaswitch MetaSphere solution.
Avaya Aura™ Session Manager performs as the SIP trunking “hub” where all inbound and
outbound SIP call routing (and other call processing) decisions are made. Avaya Aura™
Communication Manager SIP trunks and Acme Packet “session-agents” are provisioned to
terminate at Avaya Aura™ Session Manager.
The Metaswitch MetaSphere CFS solution described in these Application Notes is designed for
customers using Avaya Aura™ Communication Manager and Avaya Aura™ Session Manager.
The Metaswitch MetaSphere CFS solution provides access to service providers for local and/or
long Distance PSTN calling via standards-based SIP trunks directly, without the need for
additional TDM enterprise gateways or TDM cards and the associated maintenance costs.
1.1. Interoperability Compliance Testing
A simulated enterprise site consisting of Avaya Aura™ Communication Manager (version 5.2.1),
Avaya Aura™ Session Manager (version 5.2.1.1), Avaya Aura™ System Manager (version
5.2.1.1), and an Acme Packet 4500 Net-Net Session Director (version 6.1.0) supporting SIP
trunking was connected to the public network. The enterprise site was configured to use a
generally available SIP trunking solution provided by Metaswitch. This allowed the enterprise site
to use SIP trunking for calls to and from the PSTN.
The following features and functionality were covered during the SIP trunking interoperability
compliance testing:
•
•
•
•
•
•
•
•
Incoming calls to the enterprise site from the PSTN (using the DID numbers assigned by
Metaswitch).
Outgoing calls from the enterprise site to PSTN destinations via Metaswitch.
Calls using various analog, digital, H.323, and SIP endpoints supported by the Avaya IP
telephony solution.
Various call types including: local, long distance, and toll free calls.
Calls using various codecs.
Inbound and outbound fax calls.
DTMF tone transmission using RFC 2833 with successful voice mail navigation.
Telephone features such as hold, transfer, conference, and call forwarding.
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•
•
EC500 Features
Calls using Avaya one-X Communicator (softphone).
1.2. Support
For technical support for Metaswitch, contact your Metaswitch Networks support representative.
2. Reference Configuration
Figure 1 illustrates the reference configuration used for the DevConnect compliance testing. The
reference configuration is comprised of Avaya Customer Premises Equipment (CPE) located in the
Solution Interoperability Test Lab in Westminster, Colorado. The Avaya CPE location simulates
an enterprise customer site and uses private IP addressing. At the edge of the Avaya CPE location,
an Acme Packet Session Border Controller (SBC) provides Network Address Translation (NAT)
functionality that converts the private IP addressing to public addressing that is passed to
Metaswitch. The “inside” interface of the Acme Packet SBC is connected to a private subnet. The
“outside” interface of the Acme Packet SBC is connected to a Juniper edge router providing access
to the Metaswitch Test Lab network via the public network.
Metaswitch provided a Direct Inward Dial (DID) 10 digit number for use during the testing. The
DID was mapped by Avaya Aura™ Session Manager to an associated Avaya Aura™
Communication Manager extension.
Metaswitch used the domain 208.xxx.xxx.135. The Avaya CPE environment was assigned the
domain avaya.com. See Section 2.1 for more details regarding the domains.
The following components were used in the reference configuration and are discussed in detail in
subsequent sections.
•
•
•
•
Avaya Aura™ Session Manager on a Avaya S8800 Server
Avaya Aura™ System Manager non an Avaya S8800 Server
Avaya Aura™ Communication Manager on an Avaya S8300 Server, with an Avaya G450
Media Gateway
Avaya Aura™ Communication Manager on an Avaya S8800 Server, with an Avaya G430
Media Gateway
Note – This Communication Manager serves as a Feature Server in the reference configuration for
the Avaya 9600 Series SIP Telephones.
SIP phones (requiring advanced calling features) and non-SIP phones configured on the same
Communication Manager is currently not supported. This restriction will be lifted in future
releases of Session Manager and Communication Manager.
•
Avaya 9600 Series IP Telephones (SIP)
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•
•
•
•
•
•
•
Avaya 9600 Series IP Telephones (H.323)
Avaya one-X Communicator (H.323 softphone)
Avaya 64xx Digital Telephone
Avaya 6210 Analog Telephone
Fax Machine
Acme Packet Net-Net 4500 Session Director
Metaswitch Networks MetaSphere CFS
Figure 1: Avaya Interoperability Test Lab Reference Configuration
MJH; Reviewed:
RRR xx/xx/xxxx
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2.1. Local to Foreign Domain Conversion for Outbound Calls.
The Avaya CPE environment was assigned the domain avaya.com, and the Metaswitch domain is
208.xxx.xxx.135. For outbound calls from the Avaya CPE, the destination specified in the SIP
request URI should be 208.xxx.xxx.135. There are two methods to accomplish this.
1. Avaya Aura™ Communication Manager method – Avaya Aura™ Communication
Manager would specify the Metaswitch domain in the Far-End Domain field of the
Signaling Group form. This would result in Avaya Aura™ Communication Manager
sending a SIP request URI to Avaya Aura™ Session Manager with the following
format:
<called number>@208.xxx.xxx.135
Avaya Aura™ Session Manager would forward this URI to the Acme Packet for
transmission to Metaswitch.
2. Avaya Aura™ Session Manager method – Avaya Aura™ Communication Manager
would specify the Avaya domain (or blank) in the Far-End Domain field of the
Signaling Group form. This would result in Avaya Aura™ Communication Manager
sending a SIP request URI to Avaya Aura™ Session Manager with the following
format:
<called number>@avaya.com
By assigning an adaptation to the Acme Packet SIP Entity (see Sections 6.3.2 and
6.3.4), Avaya Aura™ Session Manager will convert the Avaya CPE domain to the
Metaswitch domain and send the following request URI to the Acme Packet:
<called number>@ 208.xxx.xxx.135
Note - In the reference configuration, method 2 was chosen.
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3. Equipment and Software Validated
The following equipment and software were used in the reference configuration.
Equipment
Avaya Aura™ Session Manager – Avaya S8800 Server w/
SM100 Board
Software/Firmware
5.2.1.1
Avaya Aura™ System Manager – Avaya S8800 Server
5.2.1.1
Avaya Aura™ Communication Manager - Avaya S8300
Server
5.2.1
with Avaya IA 770 Intuity Audix
Avaya G450 Media Gateway
-
Avaya 9600 Series IP Telephones (SIP)
Avaya 9600 Series IP Telephones (H.323)
2.5.0
3.0
Avaya one-X Communicator (H.323 softphone)
5.2
Avaya 64xx Digital Telephone
-
Avaya 6210 Analog Telephone
-
Fax Machine
Acme Packet Net-Net 4500 Session Director
Metaswitch Networks MetaSphere CFS
SCX6.1.0 MR-1 Patch 1 (WS
Build 282)
7.1.01 SU0
Table 1: Equipment and Software Used in the Reference Configuration
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4. Configure Avaya Aura™ Communication Manager for SIP
Trunking
This Section describes the steps for configuring Avaya Aura™ Communication Manager with the
necessary signaling and media characteristics for the SIP trunk connection with the Metaswitch
solution.
Note - The initial installation, configuration, and provisioning of the Avaya servers for Avaya
AuraTM Communication Manager, Avaya Media Gateways and their associated boards, as well as
the Avaya telephones, are presumed to have been previously completed and are not discussed in
these Application Notes.
The Avaya CPE site utilized an Avaya AuraTM Communication Manager running on an Avaya
S8300 server with an Avaya G450 Media Gateway.
Note – The Avaya AuraTM Communication Manager commands described in these Application
Notes were administered using the System Access Terminal (SAT). SSH was used to connect to
the SAT via the appropriate IP address, login and password.
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4.1. Verify System Capacity and Features
The Avaya AuraTM Communication Manager license file controls the customer capabilities.
Contact an authorized Avaya representative for assistance if a required feature needs to be enabled.
1. On Page 2 of the display system-parameters customer-options form, verify that the
Maximum Administered SIP Trunks value is sufficient for the combination of trunks to
the Metaswitch solution and any other SIP trunking applications. Be aware that for each
call from a non-SIP endpoint to the Metaswitch solution, one SIP trunk is used for the
duration of the call.
display system-parameters customer-options
OPTIONAL FEATURES
Page
IP PORT CAPACITIES
Maximum Administered H.323 Trunks:
Maximum Concurrently Registered IP Stations:
Maximum Administered Remote Office Trunks:
Maximum Concurrently Registered Remote Office Stations:
Maximum Concurrently Registered IP eCons:
Max Concur Registered Unauthenticated H.323 Stations:
Maximum Video Capable H.323 Stations:
Maximum Video Capable IP Softphones:
Maximum Administered SIP Trunks:
Maximum Administered Ad-hoc Video Conferencing Ports:
Maximum Number of DS1 Boards with Echo Cancellation:
Maximum TN2501 VAL Boards:
Maximum Media Gateway VAL Sources:
Maximum TN2602 Boards with 80 VoIP Channels:
Maximum TN2602 Boards with 320 VoIP Channels:
Maximum Number of Expanded Meet-me Conference Ports:
450
450
0
0
0
0
0
0
450
0
0
0
0
0
0
0
2 of
11
USED
50
3
0
0
0
0
0
0
270
0
0
0
0
0
0
0
(NOTE: You must logoff & login to effect the permission changes.)
Figure 2: System-Parameters Customer-Options Form – Page 2
Note – If any changes are made to the system-parameters customer-options form, you must log
out of the SAT and log back in for the changes to take effect.
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2. On Page 3 of the System-Parameters Customer-Options form, verify that the ARS
feature is enabled.
display system-parameters customer-options
OPTIONAL FEATURES
Abbreviated Dialing Enhanced List?
Access Security Gateway (ASG)?
Analog Trunk Incoming Call ID?
A/D Grp/Sys List Dialing Start at 01?
Answer Supervision by Call Classifier?
ARS?
ARS/AAR Partitioning?
ARS/AAR Dialing without FAC?
ASAI Link Core Capabilities?
ASAI Link Plus Capabilities?
Async. Transfer Mode (ATM) PNC?
Async. Transfer Mode (ATM) Trunking?
ATM WAN Spare Processor?
ATMS?
Attendant Vectoring?
n
n
n
n
n
y
y
y
y
y
n
n
n
n
n
Page
3 of
Audible Message Waiting?
Authorization Codes?
CAS Branch?
CAS Main?
Change COR by FAC?
Computer Telephony Adjunct Links?
Cvg Of Calls Redirected Off-net?
DCS (Basic)?
DCS Call Coverage?
DCS with Rerouting?
11
n
n
n
n
n
y
n
n
n
n
Digital Loss Plan Modification? n
DS1 MSP? n
DS1 Echo Cancellation? n
(NOTE: You must logoff & login to effect the permission changes.)
Figure 3: System-Parameters Customer-Options Form – Page 3
3. On Page 4 of the System-Parameters Customer-Options form, verify that the Enhanced
EC500, IP Trunks, and ISDN-PRI features are enabled.
display system-parameters customer-options
OPTIONAL FEATURES
Emergency Access to Attendant?
Enable 'dadmin' Login?
Enhanced Conferencing?
Enhanced EC500?
Enterprise Survivable Server?
Enterprise Wide Licensing?
ESS Administration?
Extended Cvg/Fwd Admin?
External Device Alarm Admin?
Five Port Networks Max Per MCC?
Flexible Billing?
Forced Entry of Account Codes?
Global Call Classification?
Hospitality (Basic)?
Hospitality (G3V3 Enhancements)?
IP Trunks?
y
y
n
y
n
n
n
n
n
n
n
n
n
y
n
y
Page
4 of
11
IP Stations? y
ISDN Feature Plus?
ISDN/SIP Network Call Redirection?
ISDN-BRI Trunks?
ISDN-PRI?
Local Survivable Processor?
Malicious Call Trace?
Media Encryption Over IP?
Mode Code for Centralized Voice Mail?
n
n
n
y
n
n
n
n
Multifrequency Signaling?
Multimedia Call Handling (Basic)?
Multimedia Call Handling (Enhanced)?
Multimedia IP SIP Trunking?
y
n
n
n
IP Attendant Consoles? n
(NOTE: You must logoff & login to effect the permission changes.)
Figure 4: System-Parameters Customer-Options Form – Page 4
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4.1.1 Dial Plan
In the reference configuration, five digit extensions for analog, digital, and H.323 stations were
provisioned with the format 7xxxx. Five digit extensions for SIP stations were provisioned with
the format 531xx. Trunk Access Codes (TAC) are 3 digits in length and begin with 1. The Feature
Access Code (FAC) to access ARS is one digit in length (the number “9”).
The dial plan is modified with the change dialplan analysis command.
1. On Page 1 of the form, configure the following:
• Local extensions (analog, digital, and H.323 stations):
1. In the Dialed String field, enter 7.
2. In the Total Length field, enter 5.
3. In the Call Type field, enter ext.
• Local extensions (SIP stations):
1. In the Dialed String field, enter 5.
2. In the Total Length field, enter 5.
3. In the Call Type field, enter ext.
• TAC codes:
1. In the Dialed String field, enter 1.
2. In the Total Length field, enter 3.
3. In the Call Type field, enter dac.
• FAC code – ARS access:
1. In the Dialed String field, enter 9.
2. In the Total Length field, enter 1.
3. In the Call Type field, enter fac.
change dialplan analysis
Page
DIAL PLAN ANALYSIS TABLE
Location: all
Dialed
String
1
2
4
5
7
8
9
*
Total
Length
3
5
4
5
5
5
1
3
Call
Type
dac
ext
ext
ext
ext
ext
fac
fac
Dialed
String
Total Call
Length Type
1 of
Percent Full:
Dialed
String
12
2
Total Call
Length Type
Figure 5: Change Dialplan Analysis Form – Page 1
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4.1.2 Uniform Dialplan
The uniform dial plan is modified with the change uniform-dialplan command.
1. On Page 1 of the form, configure the following:
• Local extensions (SIP stations):
1. In the Matching Pattern field, enter 531
2. In the Len field, enter 5
3. In the Del field, enter 0
4. In the Net field, enter aar
5. In the Conv field, enter n
change uniform-dialplan 0
UNIFORM DIAL PLAN TABLE
Page
1 of
2
Percent Full: 0
Matching
Pattern
531
Len Del
5
0
Insert
Digits
Node
Net Conv Num
aar n
Figure 6: Change Uniform Dialplan Form – Page 1
4.1.3 Node Names
In the IP Node Names form, verify (or assign) the node names to be used in this configuration
using the change node-names ip command.
•
•
SM2 and 10.64.20.31 are the Name and IP Address of Avaya AuraTM Session Manager.
procr and 10.64.21.41 are the Name and IP Address of the processor for Avaya AuraTM
Communication Manager.
change node-names ip
IP NODE NAMES
Name
CM-B1
CM-B2
SES-A
SM1
SM2
default
procr
IP Address
192.45.108.55
192.45.108.57
10.64.21.61
10.64.40.42
10.64.20.31
0.0.0.0
10.64.21.41
Figure 7: IP Node Names Form
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4.1.4 IP-Network-Regions
One network region was defined in the reference configuration.
The SIP trunk ip-network-regions are defined in the SIP Signaling Group form with the Far-end
Region parameter (see Section 4.1.6).
Network region assignments for ip-interfaces may be verified with the list ip-interface all
command.
list ip-interface all
IP INTERFACES
ON Type
Slot
Code/Sfx
-- ------ ----- -------y PROCR
Node Name/
IP-Address
--------------10.64.21.41
Mask
Gateway Node
Net
Rgn
---/24
--------------10.64.21.1
--1
VLAN
----
Figure 8: IP-Interface IP-Network-Region Assignments
The network-region for an ip-interface may be modified with the change ip-interface x command
where x is the board location or procr.
change ip-interface procr
Page
1 of
1
IP INTERFACES
Type: PROCR
Target socket load: 1700
Enable Interface? y
Allow H.323 Endpoints? y
Allow H.248 Gateways? y
Gatekeeper Priority: 5
Network Region: 1
IPV4 PARAMETERS
Node Name: procr
Subnet Mask: /24
Figure 9: IP-Interface IP-Network-Region Assignment
The IP-Network-Region form specifies the parameters used by the Avaya AuraTM
Communication Manager components and how components defined to different regions interact
with each other. In the reference configuration, only one ip-network region was used; however,
other combinations are possible.
Note – Avaya IP telephones inherit the ip-network-region of the procr (or C-LAN) they register to.
As a result, if an IP phone registers to the procr in the reference configuration, that phone will
become part of region 1. If an IP phone needs to be defined to a different region regardless of
registration, this may be performed with the ip-network-map command (not shown).
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4.1.4.1 IP-Network-Region 1
Ip-network-region 1 is defined for Avaya AuraTM Communication Manager components. The
network regions are modified with the change ip-network-region x command, where x is the
network region number.
1. On Page 1 of the IP Network Region form:
• Configure the Authoritative Domain field to avaya.com.
• By default, Intra-region and Inter-region IP-IP Direct Audio (media shuffling)
are set to yes to allow audio traffic to be sent directly between IP endpoints to
reduce the use of media resources.
• Set the Codec Set to 1 for the corresponding calls within the IP Network Region.
• All other values are the default values.
change ip-network-region 1
Page
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19
IP NETWORK REGION
Region: 1
Location: 1
Authoritative Domain: avaya.com
Name: Compliance Testing
MEDIA PARAMETERS
Intra-region IP-IP Direct Audio: yes
Codec Set: 1
Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048
IP Audio Hairpinning? n
UDP Port Max: 3329
DIFFSERV/TOS PARAMETERS
RTCP Reporting Enabled? y
Call Control PHB Value: 46
RTCP MONITOR SERVER PARAMETERS
Audio PHB Value: 46
Use Default Server Parameters? y
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5
AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS
RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
Figure 10: IP Network Region 1 – Page 1
4.1.5 IP Codec Sets
One IP codec set is defined in the reference configuration.
4.1.5.1 IP-Codec-Set 1
G.711MU is typically used within the same location and is often specified first. Other codecs could
be specified as well depending on local requirements. Codec set 1 is associated with ip-networkregion 1 (see Section 4.1.4.1).
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The IP-Codec-Set form is modified with the change ip-codec-set x command, where x is the
codec set number.
1. On Page 1 of the form:
• Configure the Audio Codec field 1 to G.711MU. During compliance testing
G.729B and G.729AB were also tested.
change ip-codec-set 1
Page
1 of
2
Page
2 of
2
IP Codec Set
Codec Set: 1
Audio
Codec
1: G.711MU
2:
Silence
Suppression
n
Frames
Per Pkt
2
Packet
Size(ms)
20
Figure 11: IP Codec Set 1
2. On Page 2 of the form:
• Configure the Fax field to t.38-standard.
• Configure the Fax Redundancy field to 0.
• Use the default settings for all other fields.
change ip-codec-set 1
IP Codec Set
Allow Direct-IP Multimedia? n
FAX
Modem
TDD/TTY
Clear-channel
Mode
t.38-standard
off
US
n
Redundancy
0
0
3
0
Figure 12: IP Codec Set 1 – Page 2
4.1.6 SIP Trunk Groups
SIP trunks are defined for internal calls as well as off network calls to and from the PSTN via
Metaswitch. A SIP trunk is created in Avaya AuraTM Communication Manager by provisioning a
SIP Trunk Group as well as a SIP Signaling Group.
Note – In the SIP trunk configurations below (and in the corresponding Avaya AuraTM Session
Manager configuration), TLS was selected as the transport protocol in the reference configuration.
The TCP protocol could have been used instead.
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4.1.6.1 Configure SIP Trunk for internal calls
1. Using the change signaling-group 8 command, configure the Signaling Group as follows:
•
•
Set the Group Type field to sip.
Set the Transport Method field to tls.
Note – This specifies the transport method used between Avaya AuraTM Communication Manager
and Avaya AuraTM Session Manager, not the transport method used to the Metaswitch network.
•
•
•
•
•
•
•
Specify the procr (or C-LAN) used for SIP signaling (node name procr) and the Avaya
AuraTM Session Manager (node name SM2) as the two ends of the signaling group in the
Near-end Node Name and Far-end Node Name fields, respectively. These field
values are taken from the IP Node Names form shown in Section 4.1.3.
Specify 5061 in the Near-End and Far-end Listen Port fields.
Enter the value 1 into the Far-end Network Region field. This value is for the IP
Network Region defined in Section 4.1.4.1.
Set the Far-end Domain field to avaya.com.
The Direct IP-IP Audio Connections field should be set to y to allow RTP voice paths
to be established directly between IP telephones and the Metaswitch network.
The DTMF over IP field should remain set to the default value of rtp-payload. This
value enables Avaya AuraTM Communication Manager to send DTMF tones using RFC
2833.
The default values for the other fields may be used.
change signaling-group 8
Page
1 of
1
SIGNALING GROUP
Group Number: 8
Group Type: sip
Transport Method: tls
IMS Enabled? n
Near-end Node Name: procr
Near-end Listen Port: 5061
Far-end Node Name: SM2
Far-end Listen Port: 5061
Far-end Network Region: 1
Far-end Domain: avaya.com
Incoming Dialog Loopbacks: eliminate
DTMF over IP: rtp-payload
Session Establishment Timer(min): 3
Enable Layer 3 Test? y
H.323 Station Outgoing Direct Media? n
Bypass If IP Threshold Exceeded?
RFC 3389 Comfort Noise?
Direct IP-IP Audio Connections?
IP Audio Hairpinning?
Direct IP-IP Early Media?
Alternate Route Timer(sec):
n
n
y
n
n
6
Figure 13: Internal calls SIP Trunk - Signaling Group 8
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2. Using the change trunk-group 8 command, change the Trunk Group as follows:
a. On Page 1 of the Trunk Group form:
•
Set the Group Type field to sip.
•
Choose a descriptive Group Name.
•
Specify an available trunk access code (TAC) (e.g. 108).
•
Set the Service Type field to public-ntwrk.
•
Enter 8 as the Signaling Group number.
•
Specify the Number of Members used by this SIP trunk group (e.g. 10).
change trunk-group 8
Page
1 of
21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
8
Group Type:
to SM (avaya.com)
COR:
two-way
Outgoing Display?
n
0
public-ntwrk
Auth Code?
sip
1
n
TN: 1
CDR Reports: y
TAC: 108
Night Service:
n
Signaling Group: 8
Number of Members: 10
Figure 14: Internal calls Trunk Group 8 – Page 1
b. On Page 3 of the Trunk Group form:
•
Set the Numbering Format field to public. This field specifies the format
of the calling party number sent to the far-end.
change trunk-group 8
TRUNK FEATURES
ACA Assignment? n
Page
3 of
21
Measured: none
Maintenance Tests? y
Numbering Format: public
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Show ANSWERED BY on Display? y
Figure 15: Internal calls Trunk Group 8 – Page 3
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4.1.6.2 Configure SIP Trunk for off network calls
The SIP trunk for off network calls is configured in the same fashion as the internal call SIP Trunk
except that the Far-end Domain is set to blank.
1. Using the change signaling-group 9 command, configure the Signaling Group as follows:
•
•
Set the Group Type field to sip.
Set the Transport Method field to tls.
Note – This specifies the transport method used between Avaya AuraTM Communication Manager
and Avaya AuraTM Session Manager, not the transport method used to the Metaswitch network.
•
•
•
•
•
•
•
Specify the procr (or C-LAN) used for SIP signaling (node name procr) and the Avaya
AuraTM Session Manager (node name SM2) as the two ends of the signaling group in the
Near-end Node Name and Far-end Node Name fields, respectively. These field
values are taken from the IP Node Names form shown in Section 4.1.3.
Specify 5061 in the Near-End and Far-end Listen Port fields.
Enter the value 1 into the Far-end Network Region field. This value is for the IP
Network Region defined in Section 4.1.4.1.
Leave the Far-end Domain field blank. This permits inbound calls from any foreign
domain (e.g. the Metaswitch network).
The Direct IP-IP Audio Connections field should be set to y to allow RTP voice paths
to be established directly between IP telephones and the Metaswitch network.
The DTMF over IP field should remain set to the default value of rtp-payload. This
value enables Avaya AuraTM Communication Manager to send DTMF tones using RFC
2833.
The default values for the other fields may be used.
change signaling-group 9
Page
1 of
1
SIGNALING GROUP
Group Number: 9
Group Type: sip
Transport Method: tls
IMS Enabled? n
Near-end Node Name: procr
Near-end Listen Port: 5061
Far-end Node Name: SM2
Far-end Listen Port: 5061
Far-end Network Region: 1
Far-end Domain:
Incoming Dialog Loopbacks: eliminate
DTMF over IP: rtp-payload
Session Establishment Timer(min): 3
Enable Layer 3 Test? n
H.323 Station Outgoing Direct Media? n
Bypass If IP Threshold Exceeded?
RFC 3389 Comfort Noise?
Direct IP-IP Audio Connections?
IP Audio Hairpinning?
Direct IP-IP Early Media?
Alternate Route Timer(sec):
n
n
y
n
n
6
Figure 16: Off network calls SIP Trunk - Signaling Group 9
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2. Using the change trunk-group 9 command, change the Trunk Group as follows:
a. On Page 1 of the Trunk Group form:
•
Set the Group Type field to sip.
•
Choose a descriptive Group Name.
•
Specify an available trunk access code (TAC) (e.g. 109).
•
Set the Service Type field to public-ntwrk.
•
Enter 9 as the Signaling Group number.
•
Specify the Number of Members used by this SIP trunk group (e.g. 10).
change trunk-group 9
Page
1 of
21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
9
to SM (blank)
two-way
n
0
public-ntwrk
Group Type: sip
COR: 1
Outgoing Display? n
TN: 1
CDR Reports: y
TAC: 109
Night Service:
Auth Code? n
Signaling Group: 9
Number of Members: 10
Figure 17: Off network calls Trunk Group 9 – Page 1
b. On Page 3 of the Trunk Group form:
•
Set the Numbering Format field to public. This field specifies the format
of the calling party number sent to the far-end.
change trunk-group 9
TRUNK FEATURES
ACA Assignment? n
Page
3 of
21
Measured: none
Maintenance Tests? y
Numbering Format: public
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Show ANSWERED BY on Display? y
Figure 18: Off network calls Trunk Group 9 – Page 3
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4.1.7 Public Unknown Numbering – Basic Configuration
In the reference configuration, Avaya AuraTM Communication Manager uses a 5 digit dialing plan
with extensions 7xxxx for analog, digital, and H.323 stations. The Public-Unknown-Numbering
form allows Avaya AuraTM Communication Manager to use these extensions as the calling party
number for outbound calls. Otherwise, Anonymous is displayed as the calling number. Each
extension string is defined for the trunk group(s) that the extensions may use. These trunks may be
defined individually or in contiguous ranges.
In the reference configuration, in order for a station to place off network calls (to the PSTN), the
calling number must match the DID provided by Metaswitch, or the call will be rejected by
Metaswitch. The public-unknown-numbering form was configured to convert a local calling
extension to its associated Metaswitch DID.
Use the change public-unknown-numbering x command, where x is the leading digit of the dial
plan extensions (e.g. 7).
•
•
•
•
•
Set the Ext Len field to 5.
Set the Ext Code field to 7.
Set the Trk Grp(s) field to 9.
Set the CPN Prefix field to the leading digits of the Metaswitch DID (e.g. 51021)
Set the Total CPN Len field to 10. This is the total number of digits in the DID.
With this configuration, Avaya Aura™ Communication Manager will insert 51021 for calls from a
5 digit extension (starting with digit 7), going over trunk 9. This allows the station with the
extension that matches the last 5 digits of the Metaswitch DID to place calls to the PSTN.
For internal calls:
•
•
•
•
Set the Ext Len field to 5.
Set the Ext Code field to 7.
Set the Trk Grp(s) field to 8.
Set the Total CPN Len field to 5. This is the total number of digits in the extension.
All provisioned public-unknown-numbering entries can be displayed by entering the command
display public-unknown-numbering 0 as show below.
display public-unknown-numbering 0
Page
1 of
NUMBERING - PUBLIC/UNKNOWN FORMAT
Total
Ext Ext
Trk
CPN
CPN
Len Code
Grp(s)
Prefix
Len
Total Administered: 4
5 5
5
Maximum Entries: 240
5 7
8
5
5 7
9
51021
10
2
Figure 19: Public-unknown-numbering Form – Basic Configuration
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4.1.8 Call Routing
4.1.8.1 Outbound Calls
The following sections describe the Avaya Aura™ Communication Manager provisioning required
for outbound dialing. Although Avaya Aura™ Session Manager routes all inbound and outbound
SIP trunk calls, Avaya Aura™ Communication Manager uses ARS and AAR to direct outbound
calls to Avaya Aura™ Session Manager.
4.1.8.1.1 ARS
The Automatic Route Selection feature is used to route calls via a SIP trunk, configured in Section
4.1.6.2, to Avaya Aura™ Session Manager, which in turn completes the calls to the Metaswitch. In
the reference configuration, ARS is triggered by dialing a 9 (feature access code or FAC) and then
dialing the called number. ARS matches on the called number and sends the call to a specified
route pattern.
1. Use the change feature-access-codes command to specify 9 as the access code for external
dialing.
•
Set Auto Route Selection (ARS) – Access Code 1: to 9.
change feature-access-codes
Page
1 of
FEATURE ACCESS CODE (FAC)
Abbreviated Dialing List1 Access Code:
Abbreviated Dialing List2 Access Code:
Abbreviated Dialing List3 Access Code:
Abbreviated Dial - Prgm Group List Access Code:
Announcement Access Code:
Answer Back Access Code:
Attendant Access Code:
Auto Alternate Routing (AAR) Access Code:
Auto Route Selection (ARS) - Access Code 1: 9
Access Code 2:
Automatic Callback Activation:
Deactivation:
Call Forwarding Activation Busy/DA: *11
All: *22
Deactivation: *33
Call Forwarding Enhanced Status:
Act:
Deactivation:
Call Park Access Code:
Call Pickup Access Code:
CAS Remote Hold/Answer Hold-Unhold Access Code:
CDR Account Code Access Code:
Change COR Access Code:
Change Coverage Access Code:
Conditional Call Extend Activation:
Deactivation:
Contact Closure
Open Code:
Close Code:
8
Figure 20: Feature-Access-Codes Form – Page 1
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2. Use the change ars analysis command to configure the route pattern selection rule based
upon the number dialed following the ARS access digit “9”. In the reference configuration,
outbound calls are placed to the following numbers:
•
•
1732
1303
(calls to area code 732 by dialing 9 1 732 xxx xxxx)
(calls to area code 303 by dialing 9 1 303 xxx xxxx)
For example, to specify 732 area code calls, enter the command change ars analysis 173
and enter the following values:
• Set the Dialed String field to 173.
• Set the Total Min field to 11.
• Set the Total Max field to 11.
• Set the Route Pattern field to 9 (will direct to off network calls trunk).
• Set the Type field to fnpa.
Different values may be used. These were the values used for the reference configuration.
display ars analysis 173
Page
ARS DIGIT ANALYSIS TABLE
Location: all
Dialed
String
Total
Min Max
11
11
173
Route
Pattern
9
Call
Type
fnpa
Node
Num
1 of
Percent Full:
2
2
ANI
Reqd
n
Figure 21: ARS Analysis Form
3. Using the same procedure, specify the other called number patterns in the ARS table.
4.1.8.1.2 AAR
The Automatic Alternate feature is used to route calls a the SIP trunk, configured in Section
4.1.6.1, to the Avaya Aura™ Session Manager, which in turn completes the calls to local SIP
stations. AAR matches on the called number and sends the call to a specified route pattern.
1. Use the change aar analysis command to configure the route pattern selection rule based
upon the number dialed. In the reference configuration 5 digit SIP stations were
provisioned with the extension format 531xx.
change aar analysis 531
Page
AAR DIGIT ANALYSIS TABLE
Location: all
Dialed
String
531
Total
Min Max
5
5
Route
Pattern
8
Call
Type
aar
Node
Num
1 of
Percent Full:
2
2
ANI
Reqd
n
Figure 22: AAR Analysis Form
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4.1.8.1.3
Route Patterns
The reference configuration used route-pattern 9 for ARS calls to Session Manager.
Note - Route patterns may also be used to add or delete digits prior to sending them out the
specified trunk(s). This feature was not used in the reference configuration.
1. Use the change route-pattern command to define the SIP trunk group to be used in the
route pattern that ARS selects.
•
Set the Grp No field to 9.
•
Set the FRL field to 0.
• The default values for the other fields may be used.
change route-pattern 9
Page
Pattern Number: 9
Pattern Name: Outbound-SM2
SCCAN? n
Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted
No
Mrk Lmt List Del Digits
Dgts
1: 9
0
2:
1 of
3
DCS/ IXC
QSIG
Intw
n
user
n
user
Figure 23: Route Pattern 9 – Outbound Calls to Metaswitch
2. Use the change route-pattern command to define the SIP trunk group to be used in the
route pattern that AAR selects.
•
Set the Grp No field to 8.
•
Set the FRL field to 0.
• The default values for the other fields may be used.
change route-pattern 8
Page
1 of
3
Pattern Number: 8
Pattern Name: to SIP stations
SCCAN? n
Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted
DCS/ IXC
No
Mrk Lmt List Del Digits
QSIG
Dgts
Intw
1: 8
0
n
user
2:
n
user
Figure 24: Route Pattern 8 –Calls to SIP stations
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4.1.8.2 Incoming Calls
Avaya Aura™ Session Manager is used to convert the inbound Metaswitch DID number to an
Avaya Aura™ Communication Manager extension. Therefore, no incoming digit manipulation
was required on Avaya Aura™ Communication Manager.
Note - Incoming called numbers may be changed to match a provisioned extension, if necessary,
with the Avaya Aura™ Communication Manager change inc-call-handling-trmt trunk-group x
command, where x is the receiving trunk.
4.1.9 Avaya Aura™ Communication Manager Stations (non-SIP)
In the reference configuration 5 digit non-SIP stations were provisioned with the extension format
7xxxx.
4.1.9.1 Voice Stations
The figures below show an example of an extension (Avaya H.323 IP phone). Since the phone is
an IP device, a virtual port S00027 is automatically assigned by the system. By default, three call
appearances are defined on Page 4 of the form.
1. On Page 1 of the form:
•
Set the Type field to match the station type (e.g. 9620)
•
Set the Name field to a desired value (e.g. Metaswitch)
•
Set the Security Code (optional) to a desired value (e.g. 123456)
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change station 74567
Page
1 of
5
STATION
Extension:
Type:
Port:
Name:
74567
9620
S00027
Metaswitch
Lock Messages? n
Security Code: 123456
Coverage Path 1:
Coverage Path 2:
Hunt-to Station:
BCC:
TN:
COR:
COS:
0
1
1
1
STATION OPTIONS
Loss Group: 19
Speakerphone:
Display Language:
Survivable GK Node Name:
Survivable COR:
Survivable Trunk Dest?
2-way
english
Time of Day Lock Table:
Personalized Ringing Pattern: 1
Message Lamp Ext: 74567
Mute Button Enabled? y
internal
y
Media Complex Ext:
IP SoftPhone? y
IP Video Softphone? n
Customizable Labels? y
Figure 25: Avaya H.323 IP Phone – Page 1
2. On Page 4 of the form:
• Select an empty button assignment and enter ec500. Let the timer field default to
n. This button will enable the EC500 capability on the phone (see Section 4.1.10).
• Select an empty button assignment and enter extnd-call. This button will allow a
user of this station to extend an active call to another phone number mapped to this
extension (see Section 4.1.10).
change station 74567
Page
4 of
5
STATION
SITE DATA
Room:
Jack:
Cable:
Floor:
Building:
ABBREVIATED DIALING
List1:
BUTTON ASSIGNMENTS
1: call-appr
2: call-appr
3: call-appr
Headset?
Speaker?
Mounting:
Cord Length:
Set Color:
List2:
n
n
d
0
List3:
4: ec500
Timer? n
5: extnd-call
6:
voice-mail Number:
Figure 26: Avaya H.323 IP Phone – Page 4
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4.1.10 EC500 Provisioning
The Avaya Aura™ Communication Manager EC500 feature was used to during compliance
testing. EC500 provides call coverage for an Avaya Aura™ Communication Manager station to a
second destination endpoint. Typically this endpoint is a cell phone. When EC500 is enabled on
the Avaya Aura™ Communication Manager station (by pressing the EC500 button), any inbound
call to that station will generate a new outbound call from Avaya Aura™ Communication Manager
to the provisioned EC500 destination endpoint. Similarly, if there is an existing active call at the
station, pressing the extnd-call button will generate a new outbound call from Avaya Aura™
Communication Manager to the provisioned EC500 destination endpoint.
Note – Only the basic EC500 call redirection functionality was used in the reference configuration.
EC500 supports significantly more features.
1. Use the command change off-pbx-telephone station mapping x where x is the Avaya
Aura™ Communication Manager station (e.g. 74567).
•
Station Extension – This field will automatically populate.
•
Application – Enter EC500.
•
Phone Number – Enter the phone that will also be called (e.g. 7325555555).
•
Trunk Selection – Enter 9 to route the call over trunk 9.
Note – ARS could also be entered depending on the configuration. This means ARS will be used
to determine how Avaya Aura™ Communication Manager will place the new outbound call.
•
•
Config Set – Enter 1.
Use the default values for all other fields.
change off-pbx-telephone station-mapping 74567
STATIONS WITH OFF-PBX TELEPHONE INTEGRATION
Station
Extension
74567
Application Dial
CC
Prefix
EC500
-
Phone Number
7325555555
Trunk
Selection
9
Page
1 of
Config
Set
1
3
Dual
Mode
Figure 27: EC500 Station Mapping
4.1.11 Save Avaya Aura™ Communication Manager Provisioning
Enter the save translation command to make the changes permanent.
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5. Configure Avaya Aura™ Communication Manager as a
Feature Server for SIP Trunking
This Section describes the steps for configuring Avaya Aura™ Communication Manager as a
Feature Server with the necessary signaling and media characteristics for the SIP trunk connection
with the Metaswitch solution. The Feature Server provides advanced feature capabilities to Avaya
9600 Series SIP Telephones.
Note - The initial installation, configuration, and provisioning of the Avaya servers for Avaya
AuraTM Communication Manager, Avaya Media Gateways and their associated boards, as well as
Avaya telephones, are presumed to have been previously completed and are not discussed in these
Application Notes.
The Avaya CPE site utilized Avaya AuraTM Communication Manager running on an Avaya S8800
server with an Avaya G430 Media Gateway as a Feature Server for SIP endpoints.
Note – The Avaya AuraTM Communication Manager commands described in these Application
Notes were administered using the System Access Terminal (SAT). SSH was used to connect to
the SAT via the appropriate IP address, login and password.
5.1. Verify System Capacity and Features
The Avaya AuraTM Communication Manager license file controls the customer capabilities.
Contact an authorized Avaya representative for assistance if a required feature needs to be enabled.
1. On Page 2 of the display system-parameters customer-options form, verify that the
Maximum Administered SIP Trunks is sufficient for the combination of trunks to the
Metaswitch solution and any other SIP trunking applications.
display system-parameters customer-options
OPTIONAL FEATURES
Page
IP PORT CAPACITIES
Maximum Administered H.323 Trunks:
Maximum Concurrently Registered IP Stations:
Maximum Administered Remote Office Trunks:
Maximum Concurrently Registered Remote Office Stations:
Maximum Concurrently Registered IP eCons:
Max Concur Registered Unauthenticated H.323 Stations:
Maximum Video Capable H.323 Stations:
Maximum Video Capable IP Softphones:
Maximum Administered SIP Trunks:
Maximum Administered Ad-hoc Video Conferencing Ports:
Maximum Number of DS1 Boards with Echo Cancellation:
Maximum TN2501 VAL Boards:
Maximum Media Gateway VAL Sources:
Maximum TN2602 Boards with 80 VoIP Channels:
Maximum TN2602 Boards with 320 VoIP Channels:
Maximum Number of Expanded Meet-me Conference Ports:
450
18000
0
0
0
0
0
0
300
0
0
10
0
128
128
0
2 of
11
USED
0
0
0
0
0
0
0
0
20
0
0
0
0
0
0
0
Figure 28: System-Parameters Customer-Options Form – Page 2
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Note – If any changes are made to the system-parameters customer-options form, you must log
out of the SAT and log back in for the changes to take effect.
2. On Page 3 of the System-Parameters Customer-Options form, verify that the ARS
feature is enabled.
display system-parameters customer-options
OPTIONAL FEATURES
Abbreviated Dialing Enhanced List?
Access Security Gateway (ASG)?
Analog Trunk Incoming Call ID?
A/D Grp/Sys List Dialing Start at 01?
Answer Supervision by Call Classifier?
ARS?
ARS/AAR Partitioning?
ARS/AAR Dialing without FAC?
ASAI Link Core Capabilities?
ASAI Link Plus Capabilities?
Async. Transfer Mode (ATM) PNC?
Async. Transfer Mode (ATM) Trunking?
ATM WAN Spare Processor?
ATMS?
Attendant Vectoring?
n
n
n
n
n
y
y
y
y
y
n
n
n
n
n
Page
3 of
Audible Message Waiting?
Authorization Codes?
CAS Branch?
CAS Main?
Change COR by FAC?
Computer Telephony Adjunct Links?
Cvg Of Calls Redirected Off-net?
DCS (Basic)?
DCS Call Coverage?
DCS with Rerouting?
11
n
n
n
n
n
y
n
n
n
n
Digital Loss Plan Modification? n
DS1 MSP? n
DS1 Echo Cancellation? n
(NOTE: You must logoff & login to effect the permission changes.)
Figure 29: System-Parameters Customer-Options Form – Page 3
3. On Page 4 of the System-Parameters Customer-Options form, verify that the IP Trunks
feature is enabled.
display system-parameters customer-options
OPTIONAL FEATURES
Emergency Access to Attendant?
Enable 'dadmin' Login?
Enhanced Conferencing?
Enhanced EC500?
Enterprise Survivable Server?
Enterprise Wide Licensing?
ESS Administration?
Extended Cvg/Fwd Admin?
External Device Alarm Admin?
Five Port Networks Max Per MCC?
Flexible Billing?
Forced Entry of Account Codes?
Global Call Classification?
Hospitality (Basic)?
Hospitality (G3V3 Enhancements)?
IP Trunks?
y
y
y
y
n
n
n
n
n
n
n
n
n
y
n
y
Page
4 of
11
IP Stations? y
ISDN Feature Plus?
ISDN/SIP Network Call Redirection?
ISDN-BRI Trunks?
ISDN-PRI?
Local Survivable Processor?
Malicious Call Trace?
Media Encryption Over IP?
Mode Code for Centralized Voice Mail?
n
n
n
y
n
n
n
n
Multifrequency Signaling?
Multimedia Call Handling (Basic)?
Multimedia Call Handling (Enhanced)?
Multimedia IP SIP Trunking?
y
n
n
n
Figure 30: System-Parameters Customer-Options Form – Page 4
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5.1.1 Dial Plan
In the reference configuration, five digit extensions for analog, digital, and H.323 stations were
provisioned with the format 7xxxx. Five digit extensions for SIP stations were provisioned with
the format 531xx. Trunk Access Codes (TAC) are 3 digits in length and begin with 1. The Feature
Access Code (FAC) to access ARS is one digit in length (the number “9”).
The dial plan is modified with the change dialplan analysis command.
1. On Page 1 of the form:
• Local extensions (analog, digital, and H.323 stations):
1. In the Dialed String field enter 7.
2. In the Total Length field enter 5.
3. In the Call Type field enter ext.
• Local extensions (SIP stations):
1. In the Dialed String field enter 5.
2. In the Total Length field enter 5.
3. In the Call Type field enter ext.
• TAC codes:
1. In the Dialed String field enter 1.
2. In the Total Length field enter 3.
3. In the Call Type field enter dac.
• FAC code – ARS access:
1. In the Dialed String field enter 9.
2. In the Total Length field enter 1.
3. In the Call Type field enter fac.
change dialplan analysis
Page
DIAL PLAN ANALYSIS TABLE
Location: all
Dialed
String
1
2
4
5
6
7
8
9
Total
Length
3
5
4
5
5
5
5
1
Call
Type
dac
ext
ext
ext
aar
ext
aar
fac
Dialed
String
Total Call
Length Type
1 of
Percent Full:
Dialed
String
12
1
Total Call
Length Type
Figure 31: Change Dialplan Analysis Form – Page 1
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5.1.2 Uniform Dialplan
The uniform dial plan is modified with the change uniform-dialplan command.
1. On Page 1 of the form, configure the following:
• Local extensions (SIP stations):
1. In the Matching Pattern field, enter 7
2. In the Len field, enter 5
3. In the Del field, enter 0
4. In the Net field, enter aar
5. In the Conv field, enter n
change uniform-dialplan 0
UNIFORM DIAL PLAN TABLE
Page
1 of
2
Percent Full: 0
Matching
Pattern
7
Len Del
5
0
Insert
Digits
Node
Net Conv Num
aar n
Figure 32: Change Uniform Dialplan Form – Page 1
5.1.3 Node Names
In the IP Node Names form, verify (or assign) the node names to be used in this configuration
using the change node-names ip command.
•
•
SM01 and 10.64.20.31 are the Name and IP Address of Avaya AuraTM Session Manager.
procr and 10.64.20.25 are the Name and IP Address of the processor for Avaya AuraTM
Communication Manager.
display node-names ip
IP NODE NAMES
Name
SM01
default
procr
IP Address
10.64.20.31
0.0.0.0
10.64.20.25
Figure 33: IP Node Names Form
5.1.4 IP-Network-Regions
One network region was defined in the reference configuration.
The SIP trunk ip-network-regions are defined in the SIP Signaling Group form with the Far-end
Region parameter (see Section 4.1.6).
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Network region assignments for ip-interfaces may be verified with the list ip-interface all
command.
list ip-interface all
IP INTERFACES
ON Type
Slot
Code/Sfx
-- ------ ----- -------y PROCR
Node Name/
IP-Address
--------------10.64.20.25
Mask
Gateway Node
Net
Rgn
---/24
--------------10.64.20.1
--1
VLAN
----
Figure 34: IP-Interface IP-Network-Region Assignments
The network-region for an ip-interface may be modified with the change ip-interface x command
where x is the board location or procr.
change ip-interface procr
Page
1 of
1
IP INTERFACES
Type: PROCR
Target socket load: 1700
Enable Interface? y
Allow H.323 Endpoints? y
Allow H.248 Gateways? y
Gatekeeper Priority: 5
Network Region: 1
IPV4 PARAMETERS
Node Name: procr
Subnet Mask: /24
Figure 35: IP-Interface IP-Network-Region Assignment.
The IP-Network-Region form specifies the parameters used by the Avaya AuraTM
Communication Manager components and how components defined to different regions interact
with each other. In the reference configuration, only one ip-network region was used; however,
other combinations are possible.
Note – Avaya IP telephones inherit the ip-network-region of the procr (or C-LAN) they register to.
As a result, if an IP phone registers to the procr in the reference configuration, that phone will
become part of region 1. If an IP phone needs to be defined to a different region regardless of
registration, this may be performed with the ip-network-map command.
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5.1.4.1 IP-Network-Region 1
Ip-network-region 1 is defined for Avaya AuraTM Communication Manager components. The
network regions are modified with the change ip-network-region x command, where x is the
network region number.
1. On Page 1 of the IP Network Region form:
• Configure the Authoritative Domain field to avaya.com.
• By default, Intra-Region and Inter-Region IP-IP Direct Audio (media shuffling)
is set to yes to allow audio traffic to be sent directly between IP endpoints to reduce
the use of media resources.
• Set the Codec Set to 1 for the corresponding calls within the IP Network Region.
• All other values are the default values.
change ip-network-region 1
Page
1 of
19
IP NETWORK REGION
Region: 1
Location: 1
Authoritative Domain: avaya.com
Name:
MEDIA PARAMETERS
Intra-region IP-IP Direct Audio: yes
Codec Set: 1
Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048
IP Audio Hairpinning? n
UDP Port Max: 3329
DIFFSERV/TOS PARAMETERS
RTCP Reporting Enabled? y
Call Control PHB Value: 46
RTCP MONITOR SERVER PARAMETERS
Audio PHB Value: 46
Use Default Server Parameters? y
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5
AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS
RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
Figure 36: IP Network Region 1 – Page 1
5.1.5 IP Codec Sets
One codec set is defined in the reference configuration.
5.1.5.1 IP-Codec-Set 1
G.711MU is typically used within the same location and is often specified first. Other codecs could
be specified as well depending on local requirements. Codec set 1 is associated with ip-networkregion 1 (see Section 5.1.4.1).
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The IP-Codec-Set form is modified with the change ip-codec x command, where x is the codec
set number.
1. On Page 1 of the form:
• Configure the Audio Codec field 1 to G.711MU. During compliance testing
G.729B and G.729AB were also tested.
change ip-codec-set 1
Page
1 of
2
Page
2 of
2
IP Codec Set
Codec Set: 1
Audio
Codec
1: G.711MU
2:
Silence
Suppression
n
Frames
Per Pkt
2
Packet
Size(ms)
20
Figure 37: IP Codec Set 1
2. On Page 2 of the form:
• Configure the Fax field to t.38-standard.
• Configure the Fax Redundancy field to 0.
• Use the default settings for all other fields.
change ip-codec-set 1
IP Codec Set
Allow Direct-IP Multimedia? n
FAX
Modem
TDD/TTY
Clear-channel
Mode
t.38-standard
off
US
n
Redundancy
0
0
3
0
Figure 38: IP Codec Set 1 – Page 2
5.1.6 SIP Trunk Groups
SIP trunks are defined for internal calls as well as network calls to and from the PSTN via
Metaswitch. A SIP trunk is created in Avaya AuraTM Communication Manager by provisioning a
SIP Trunk Group as well as a SIP Signaling Group.
Note – In the SIP trunk configurations below (and in the corresponding Avaya AuraTM Session
Manager configuration), TCP was selected as the transport protocol in the reference configuration.
The TLS protocol could have been used instead.
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5.1.6.1 Configure SIP Trunk for internal calls
1. Using the change signaling-group 1 command, configure the Signaling Group as follows:
•
•
Set the Group Type field to sip.
Set the Transport Method field to tcp.
Note – This specifies the transport method used between Avaya AuraTM Communication Manager
and Avaya AuraTM Session Manager, not the transport method used to the Metaswitch network.
•
•
•
•
•
•
•
•
Set the IMS Enabled? field to y.
Specify the procr (or C-LAN) used for SIP signaling (node name procr) and the Avaya
AuraTM Session Manager (node name SM01) as the two ends of the signaling group in
the Near-end Node Name and Far-end Node Name fields, respectively. These field
values are taken from the IP Node Names form shown in Section 5.1.3.
Specify 5060 in the Near-End and Far-end Listen Port fields.
Enter the value 1 into the Far-end Network Region field. This value is for the IP
Network Region defined in Section 5.1.4.1.
Set the Far-end Domain field to avaya.com.
The Direct IP-IP Audio Connections field should be set to y to allow RTP voice paths
to be established directly between IP telephones and the Metaswitch network.
The DTMF over IP field should remain set to the default value of rtp-payload. This
value enables Avaya AuraTM Communication Manager to send DTMF tones using RFC
2833.
The default values for the other fields may be used.
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change signaling-group 1
Page
1 of
1
SIGNALING GROUP
Group Number: 1
Group Type: sip
Transport Method: tcp
IMS Enabled? y
Near-end Node Name: procr
Near-end Listen Port: 5060
Far-end Node Name: SM01
Far-end Listen Port: 5060
Far-end Network Region: 1
Far-end Domain: avaya.com
Incoming Dialog Loopbacks: eliminate
DTMF over IP: rtp-payload
Session Establishment Timer(min): 3
Enable Layer 3 Test? y
H.323 Station Outgoing Direct Media? n
Bypass If IP Threshold Exceeded?
RFC 3389 Comfort Noise?
Direct IP-IP Audio Connections?
IP Audio Hairpinning?
Direct IP-IP Early Media?
Alternate Route Timer(sec):
n
n
y
n
n
6
Figure 39: Internal calls SIP Trunk - Signaling Group 1
2. Using the change trunk-group 1 command, change the Trunk Group as follows:
a. On Page 1 of the Trunk Group form:
•
Set the Group Type field to sip.
•
Choose a descriptive Group Name.
•
Specify an available trunk access code (TAC) (e.g. 101).
•
Set the Service Type field to tie.
•
Enter 1 as the Signaling Group number.
•
Specify the Number of Members used by this SIP trunk group (e.g. 10).
change trunk-group 1
Page
1 of
21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
1
Group Type:
to SM (avaya.com)
COR:
two-way
Outgoing Display?
n
0
tie
Auth Code?
sip
1
n
TN: 1
CDR Reports: y
TAC: 101
Night Service:
n
Signaling Group: 1
Number of Members: 10
Figure 40: Internal calls Trunk Group 1 – Page 1
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b. On Page 3 of the Trunk Group form:
•
Set the Numbering Format field to private. This field specifies the format
of the calling party number sent to the far-end.
change trunk-group 1
TRUNK FEATURES
ACA Assignment? n
Page
3 of
21
Measured: none
Maintenance Tests? y
Numbering Format: private
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Show ANSWERED BY on Display? y
Figure 41: Internal calls Trunk Group 1 – Page 3
5.1.6.2 Configure SIP Trunk for off network calls
The SIP trunk for off network calls is configured in the same fashion as the internal call SIP Trunk
except that the Far-end Domain is set to blank.
1. Using the change signaling-group 2 command, configure the Signaling Group as follows:
•
•
Set the Group Type field to sip.
Set the Transport Method field to tcp.
Note – This specifies the transport method used between Avaya AuraTM Communication Manager
and Avaya AuraTM Session Manager, not the transport method used to the Metaswitch network.
•
•
•
•
•
•
Set the IMS Enabled? field to y.
Specify the procr (or C-LAN) used for SIP signaling (node name procr) and the Avaya
AuraTM Session Manager (node name SM01) as the two ends of the signaling group in
the Near-end Node Name and Far-end Node Name fields, respectively. These field
values are taken from the IP Node Names form shown in Section 5.1.3.
Specify 5060 in the Near-End and Far-end Listen Port fields.
Enter the value 1 into the Far-end Network Region field. This value is for the IP
Network Region defined in Section 5.1.4.1.
Leave the Far-end Domain field blank. This permits inbound calls from any foreign
domain (e.g. the Metaswitch network).
The Direct IP-IP Audio Connections field should be set to y to allow RTP voice paths
to be established directly between IP telephones and the Metaswitch network.
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The DTMF over IP field should remain set to the default value of rtp-payload. This
value enables Avaya AuraTM Communication Manager to send DTMF tones using RFC
2833.
• The default values for the other fields may be used.
•
change signaling-group 2
Page
1 of
1
SIGNALING GROUP
Group Number: 2
Group Type: sip
Transport Method: tcp
IMS Enabled? y
Near-end Node Name: procr
Near-end Listen Port: 5060
Far-end Node Name: SM01
Far-end Listen Port: 5060
Far-end Network Region: 1
Far-end Domain:
Incoming Dialog Loopbacks: eliminate
DTMF over IP: rtp-payload
Session Establishment Timer(min): 3
Enable Layer 3 Test? n
H.323 Station Outgoing Direct Media? n
Bypass If IP Threshold Exceeded?
RFC 3389 Comfort Noise?
Direct IP-IP Audio Connections?
IP Audio Hairpinning?
Direct IP-IP Early Media?
Alternate Route Timer(sec):
n
n
y
n
n
6
Figure 42: Off network calls SIP Trunk - Signaling Group 2
2. Using the change trunk-group 2 command, change the Trunk Group as follows:
a. On Page 1 of the Trunk Group form:
•
Set the Group Type field to sip.
•
Choose a descriptive Group Name.
•
Specify an available trunk access code (TAC) (e.g 102).
•
Set the Service Type field to tie.
•
Enter 2 as the Signaling Group number.
•
Specify the Number of Members used by this SIP trunk group (e.g. 10).
change trunk-group 2
Page
1 of
21
TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
2
to SM (blank)
two-way
n
0
tie
Group Type: sip
COR: 1
Outgoing Display? n
TN: 1
CDR Reports: y
TAC: 102
Night Service:
Auth Code? n
Signaling Group: 2
Number of Members: 10
Figure 43: Off network calls Trunk Group 2 – Page 1
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b. On Page 3 of the Trunk Group form:
•
Set the Numbering Format field to private. This field specifies the format
of the calling party number sent to the far-end.
change trunk-group 2
TRUNK FEATURES
ACA Assignment? n
Page
3 of
21
Measured: none
Maintenance Tests? y
Numbering Format: private
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Show ANSWERED BY on Display? y
Figure 44: Off network calls Trunk Group 2 – Page 3
5.1.7 Private Unknown Numbering – Basic Configuration
In the reference configuration, Avaya AuraTM Communication Manager uses a 5 digit dialing plan
with extensions 531xx for SIP stations. The Private-Unknown-Numbering form allows Avaya
AuraTM Communication Manager to use these extensions as the calling party number for outbound
calls. Otherwise, Anonymous is displayed as the calling number. Each extension string is defined
for the trunk group(s) that the extensions may use. These trunks may be defined individually or in
contiguous ranges.
Use the change private-unknown-numbering x command, where x is the leading digit of the dial
plan extensions (e.g. 5).
•
•
•
•
Set the Ext Len field to 5.
Set the Ext Code field to 5.
Set the Trk Grp(s) field to 1.
Set the Total CPN Len field to 5
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All provisioned private-unknown-numbering entries can be displayed by entering the command
display private-unknown-numbering 0 as show below.
display private-numbering 0
Page
1 of
2
NUMBERING - PRIVATE FORMAT
Ext
Len
4
4
5
Ext
Code
4
4
5
Trk
Grp(s)
1
2
1
Private
Prefix
Total
Len
4
Total Administered: 3
4
Maximum Entries: 540
5
Figure 45: Private-unknown-numbering Form – Basic Configuration
5.1.8 Call routing
5.1.8.1 Internal Calls
The following sections describe the Avaya Aura™ Communication Manager provisioning required
for dialing internal non-SIP extensions.
Note –Metaswitch only assigned one DID number that had access to the PSTN. The configuration
required for inbound and outbound PSTN calls to and from an H.323 station on Avaya Aura™
Communication Manager was shown in Section 4.1.8. Although, not shown here, similar
administration can be done for a SIP station on Avaya Aura™ Communication Manager as a
Feature Server.
5.1.8.1.1 AAR
The Automatic Alternate feature is used to route calls via a SIP trunk, configured in Section
5.1.6.1, to the Avaya Aura™ Session Manager, which in turn completes the calls to local stations.
AAR matches on the called number and sends the call to a specified route pattern.
1. Use the change aar analysis command to configure the route pattern selection rule based
upon the number dialed. In the reference configuration, calls are placed to non-SIP stations
with 5 digit extensions (7xxxx).
change aar analysis 7
Page
AAR DIGIT ANALYSIS TABLE
Location: all
Dialed
String
7
Total
Min Max
5
5
Route
Pattern
1
Call
Type
aar
Node
Num
1 of
Percent Full:
2
1
ANI
Reqd
n
Figure 46: AAR Analysis Form
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5.1.8.1.2
Route Patterns
The reference configuration used route-pattern 1 for internal calls.
Note - Route patterns may also be used to add or delete digits prior to sending them out the
specified trunk(s). This feature was not used in the reference configuration.
1. Use the change route-pattern command to define the SIP trunk group included in the
route pattern that AAR selects.
•
Set the Grp No field to 1.
•
Set the FRL field to 0.
• Let all other parameters default.
change route-pattern 1
Pattern Number: 1
Pattern Name: to SM
SCCAN? n
Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted
No
Mrk Lmt List Del Digits
Dgts
1: 1
0
2:
Page
1 of
3
DCS/ IXC
QSIG
Intw
n
user
n
user
Figure 47: Route Pattern 1 – Internal Calls
5.1.9 Save Avaya Aura™ Communication Manager Provisioning
Enter the save translation command to make the changes permanent.
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6. Avaya Aura™ Session Manager Provisioning
This section provides the procedures for configuring Avaya Aura™ Session Manager as
provisioned in the reference configuration. Avaya Aura™ Session Manager is comprised of two
functional components: the Avaya Aura™ Session Manager server and the System Manager
management server. All SIP call provisioning for Avaya Aura™ Session Manager is performed via
the System Manager web interface and is then downloaded into Avaya Aura™ Session Manager.
Note – The following sections assume that Avaya Aura™ Session Manager and System Manager
have been installed and that network connectivity exists between the two platforms.
6.1. Network Interfaces
Avaya Aura™ Session Manager is comprised of two main components, the server itself and the
SM-100 card.
The Avaya Aura™ Session Manager SM-100 card has four network interface ports, with one being
the connection to the SIP VoIP network. This interface is used for all inbound and outbound SIP
signaling and must have network connectivity to all provisioned SIP Entities.
The Avaya Aura™ Session Manager server has two network interface ports with one being the
port used for management/provisioning of Avaya Aura™ Session Manager. This port must have
network connectivity to System Manager.
Note –In the reference configuration the SM-100 interface and the Avaya Aura™ Session
Manager server interface were both connected to the same IP network. If desired, the System
Manager/Avaya Aura™ Session Manager management connection may use a different network
than the SM-100 connection.
6.2. Logging into System Manager
The following provisioning is performed via System Manager to enable SIP trunking:
•
Network Routing Policy
o SIP Domains - Define domains that may send calls to Avaya Aura™ Session
Manager.
o Locations – Logical/physical areas that may be occupied by SIP Entities
o SIP Entities – Typically devices corresponding to the SIP telephony systems
including Avaya Aura™ Session Manager itself, however they may includes other
devices such as SBCs.
o Entity Links – Connection information which define the SIP trunk parameters used
by Avaya Aura™ Session Manager when routing calls to/from other SIP Entities.
o Dial Patterns – Matching digit patterns which govern to which SIP Entity a call is
routed.
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•
o Routing Policies - Policies that determine which control call routing between the
SIP Entities based on applicable Dial Patterns.
o Time Ranges – Specified windows during which SIP call processing is permitted
for a particular Routing Policies.
Avaya Aura™ Session Manager – Information corresponding to the Avaya Aura™
Session Manager Server to be managed by System Manager.
Configuration is accomplished by accessing the browser-based GUI of System Manager, using the
URL https://<ip-address>/SMGR, where “<ip-address>” is the IP address of System Manager.
Log in with the appropriate credentials.
Figure 48: System Manager GUI Log On Screen
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6.3. Network Routing Policy
After logging in, expand the Network Routing Policy Link on the left side as shown.
Figure 49: Network Routing Policy Menu
6.3.1 SIP Domains
In the reference configuration, one SIP domain was used. The Avaya CPE location domain is
avaya.com.
1.
2.
3.
4.
5.
Select SIP Domains from the menu.
Select New.
Enter the SIP Domain (avaya.com) in the Name field.
Enter a description in the Notes field if desired.
Click on the Commit button.
Note – On most of the following forms, to edit or delete an entry, click the box next to the item to
select it and to make the Edit and Delete buttons available.
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Figure 50: SIP Domain Menu
6.3.2 Adaptations
Avaya Aura™ Session Manager provides for specialized code modules to process specific call
processing requirements of various vendors and/or services. These modules are called adaptations.
6.3.2.1 DigitConversionAdapter
This adaptation allows Avaya Aura™ Session Manager to convert inbound and/or outbound digits
in SIP Request-URI, History-Info header, P-Asserted-Identity header, and Notify messages, based
on the SIP Entities to which this adaptation is defined. This functionality is similar to the Avaya
Aura™ Communication Manager public-unknown-numbering and incoming-call-handlingtreatment capabilities.
Avaya Aura™ Session Manager will perform digit conversion based on whether the digits are
being received (incoming) or sent (outgoing) by Avaya Aura™ Session Manager with another SIP
Entity. For example, on a call from Avaya Aura™ Communication Manager to Metaswitch, the
call leg from Avaya Aura™ Communication Manager to Avaya Aura™ Session Manager is
incoming, while the call leg from Avaya Aura™ Session Manager to the Acme Packet is outgoing.
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1. Select Adaptations from the menu.
2. Select New.
•
Enter a descriptive name (e.g. Metaswitch).
•
Specify DigitConversionAdapter in the Adaptation Module field.
•
Set Module parameter to the domain of Metaswitch. The reference configuration
required that domain contained in the Request URI to be replaced with the
Metaswitch domain before being sent out to Metaswitch via the Acme Packet. The
domain replacement was performed by specifying the Metaswitch domain here.
•
Leave the Egress URI Parameters field blank (this is for adding additional
parameters such as user=phone).
•
Enter a description in the Notes field if desired.
For outgoing calls from Avaya Aura™ Communication Manager to the PSTN, extension 74565 is
converted to the Metaswitch DID 5102174567 via the public-unknown-numbering form on
Communication Manager (see Section 4.1.7).
For incoming calls, the Metaswitch DID 5102174567 is converted to Avaya Aura™
Communication Manager extension 74567 via this adaptation.
3. Click the Add button and enter:
•
Matching Pattern – The digit string to match 5102174567
•
Min – The minimum number of digits 10
•
Max – The maximum number of digits 10
•
Delete Digits – The number of digits to delete 5
•
Insert Digits – The digit to be inserted 0
•
Address to Modify - origination/destination/both – Associated headers to be
monitored for matching digits. Both
•
Notes - Enter a description in the Notes field if desired.
4. When completed, the Adaptation Details window for DigitConversionAdapter will look
like Figure 51.
5. Click on the Commit button.
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Figure 51: DigitConversionAdapter Adaptation
6.3.3 Locations
Locations are used to identify logical and/or physical locations where SIP Entities reside, by
specifying the IP addressing for the locations as well as for purposes of bandwidth management if
required. In the reference configuration, only the Avaya CPE site was defined as a Location.
To add a Location, select Locations in the left Network Routing Policy menu and click on the
New button on the right. The screen shown below will open.
Enter a descriptive Location name in the Name field (e.g. 10.64.20/21.0).
Enter a description in the Notes field if desired.
Under the Location Pattern heading, click on Add.
Enter the IP address information for the Location (e.g. 10.64.20.*)
Enter a description in the Notes field if desired.
Repeat steps 3 thru 5 if the Location has multiple IP segments.
Modify the remaining values on the form, if necessary; otherwise, use all the default
values.
8. Click on the Commit button.
9. Repeat all the steps for each new Location.
1.
2.
3.
4.
5.
6.
7.
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Figure 52: Locations Menu
6.3.4 SIP Entities
A SIP Entity must be added for Avaya Aura™ Session Manager and for each network component
that has a SIP trunk provisioned to Avaya Aura™ Session Manager. In the reference configuration
the SIP Entities are provisioned for:
•
•
•
•
Avaya Aura™ Communication Manager
Avaya Aura™ Communication Manager (Feature Server)
Acme Packet
Avaya Aura™ Session Manager itself.
To add a SIP Entity, select SIP Entities on the left Network Routing Policy menu and click on
the New button on the right. The screen shown below is displayed.
1. General Section
a. Enter a descriptive Location name in the Name field.
b. Enter the IP address for the SIP Entity.
c. From the Type drop down menu select a type that best matches the SIP Entity (e.g.
CM).
d. Enter a description in the Notes field if desired.
e. From the Adaptations drop down menu, select the adaptation required for this
Entity (see Section 6.3.2).
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For the Acme Packet Entity, the Metaswitch adaptation is selected. This
function is applied to convert the Metaswitch DID to a Communication
Manager extension. It also convert the outbound call (Avaya Aura™
Session Manager to Acme) request URI domain from the Avaya CPE
domain, used by Avaya Aura™ Communication Manager, to the
Metaswitch domain.
f. From the Locations drop down menu, select 10.64.20/21.0.
g. Select the appropriate time zone.
h. Accept the other default values.
Sip Link Monitoring section
a. Select the desired option.
Port section
a. When defining a SIP Entity for Avaya Aura™ Session Manager itself and SM is
selected from the Type drop down menu, an additional section called Ports will
appear. Click Add, then edit the fields in the resulting new row:
•
Enter the Port number on which the system listens for SIP requests.
•
Select the transport Protocol to be used.
•
Select the SIP Domain configured in Section 6.3.1 for the Default Domain.
b. Repeat step 3 for each Port to be configured.
Click on Commit.
Repeat these steps for each SIP Entity.
•
2.
3.
4.
5.
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Figure 53: Session Manager SIP Entity Details
The following SIP Entity values were specified in the reference configuration.
Name
IP Address
Type
Adaptation
Location
Port
Protocol
S8300_G450_AE
CM8800_G430_FS
Acme Packet
SM 01
10.64.21.41
10.64.20.25
10.64.20.106
10.64.20.31
CM
CM
Other
SM
Metaswitch
-
10.64.20/21.0
10.64.20/21.0
10.64.20/21.0
10.64.20/21.0
5060
5060
5061
TCP
UDP
TLS
Default
Domain
avaya.com
Table 2: SIP Entities Provisioning
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6.3.5 Entity Links
Entity Links defined the connections between the SIP Entities and Avaya Aura™ Session
Manager. In the reference configuration, Entity Links are defined between Avaya Aura™ Session
Manager and:
•
•
•
The Avaya Aura™ Communication Manager
The Avaya Aura™ Communication Manager (Feature Server)
Acme Packet
To add an Entity Link, select Entity Links on the left Network Routing Policy menu and click on
the New button on the right. The screen shown below is displayed.
1. Enter a descriptive name in the Name field.
2. In the SIP Entity 1 drop down menu, select the Avaya Aura™ Session Manager SIP Entity
created in Section 6.3.4 (e.g. SM 01).
3. In the Port field, enter the port to be used (e.g. 5060 or 5061).
4. In the SIP Entity 2 drop down menu, select the one of the three entities in the bullet list
above (which were created in Section 6.3.4).
5. In the Port field, enter the port to be used (e.g. 5060 or 5061).
6. Check the Trusted box.
7. In the Protocol drop down menu, select the protocol to be used.
8. Enter a description in the Notes field if desired (not shown).
9. Click on the Commit button.
10. Repeat steps 1 – 9 for each Entity Link.
The following Entity Links were specified in the reference configuration.
Name
SIP Entity 1
Protocol
Port
SIP Entity 2
Port
SM 01_S8300_G450_AE_5061_TLS
SM 01_CM8800_FS_5060_TCP
SM 01_Acme_Packet_5060_TCP
SM 01
SM 01
SM 01
TLS
TCP
TCP
5061
5060
5060
S8300_G450_AE
CM8800_G430_FS
Acme Packet
5061
5060
5060
Table 3: Entity Link Provisioning
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Figure 54: Entity Link – Communication Manager
6.3.6 Time Ranges
The Time Ranges form allows admission control criteria to be specified for Routing Policies
(Section 6.3.7). In the reference configuration, no restrictions were used.
To add a Time Range, select Time Ranges on the left Network Routing Policy menu and click on
the New button on the right. The screen shown below is displayed.
1.
2.
3.
4.
5.
6.
Enter a descriptive Location name in the Name field (e.g. Anytime).
Check each day of the week.
In the Start Time field, enter 00:00.
In the End Time field, enter 23:59.
Enter a description in the Notes field if desired.
Click the Commit button.
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Figure 55: Time Ranges
6.3.7 Routing Policies
Routing Policies associate destination SIP Entities (Section 6.3.4) with Time of Day admission
control parameters (Section 6.3.6) and Dial Patterns (Section 6.3.8). In the reference
configuration, Routing Policies are defined for:
•
•
Inbound calls to Avaya Aura™ Communication Manager
Outbound calls to the Metaswitch network
Note – Since the SIP endpoints in the reference configuration register with Avaya Aura™ Session
Manager, Session Manager knows how to route calls to those extensions and it is not necessary to
create a routing policy for Avaya Aura™ Communication Manager (Feature Server).
Note – In the reference configuration the Regular Expressions parameters was not used.
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Name
SIP Entity as
Destination
Time Of
Day
Dial
Pattern(s)
to S8300_G450_AE
S8300_G450_AE
Anytime
7xxxx
to Acme_Packet
Acme Packet
Anytime
303xxxxxxx
732xxxxxxx
Notes
Any call to a 5 digit
extension beginning with
7 will be routed to Avaya
Aura™ Communication
Manager
Any call to a 10 digit
number beginning with
303 or 732 will be routed
to Acme Packet
Table 4: Routing Policy Provisioning
To add a Routing Policy, select Routing Policies on the left Network Routing Policy menu and
click on the New button on the right. The Routing Policy Details window will open.
1. General section
a. Enter a descriptive name in the Name field.
b. Enter a description in the Notes field if desired.
2. SIP Entity as Destination section
a. Click the Select button.
b. Select the SIP Entity that will be the destination for this call.
c. Click the Select button and return to the Routing Policy Details form.
3. Time of Day section
a. Leave default values.
Note – Multiple time ranges may be selected and a Ranking value applied (0 is the
highest).
4. Dial Pattern section
Note – Step 4 may be skipped. Dial Patterns will be mapped to Routing Policies in the
Section 6.3.8.
a. Click the Add button and select the Dial Pattern for this Routing Policy.
b. Click on Select and return to the Routing Policy Details form.
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Figure 56: Routing Policy Details
5. Click the Commit button.
6. Repeat steps 1 thru 5 for each Routing Policy.
7. Click the Commit button.
6.3.8 Dial Patterns
Dial Patterns define digit strings to be matched for inbound and outbound calls. In addition, the
domain in the request URI is also examined.
Note – The Dial Pattern digit string with the most complete match will be selected. For example if
the 5 digit string 700 is defined first in the list, and the 5 digit string 70001 is defined last, a call for
70001 will match on the 70001 string.
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The following Dial Patterns were provisioned in the reference configuration.
Pattern
Min
Max
SIP Domain
Originating Location
Routing Policy Name
7
303
5
10
5
10
avaya.com
avaya.com
ALL
10.64.20/21.0
to S8300_G450_AE
to Acme_Packet
732
10
10
avaya.com
10.64.20/21.0
to Acme_Packet
Table 5: Routing Policy Provisioning
Note – The Metaswitch adaptation is provisioned on the Acme Packet SIP Entity. This means that
the conversion from the Metaswitch DID to the Avaya Aura™ Communication Manager extension
is performed before the dial pattern match for inbound calls.
To add a Dial Pattern, select Dial Patterns on the left Network Routing Policy menu and click on
the New button on the right. The screen shown below is displayed. In this example, a Request URI
to a 10 digit number beginning with 732xxxxxxx, and sent by avaya.com, is defined (this would be
an outbound call from Avaya Aura™ Communication Manager to Avaya Aura™ Session
Manager, destined for Metaswitch).
1. General section
a. Enter a unique pattern in the Pattern field (e.g. 732).
b. In the Min column enter the minimum number of digits (e.g. 10).
c. In the Max column enter the maximum number of digits (e.g. 10).
d. In the SIP Domain field drop down menu select the domain that will be contained
in the Request URI received by Avaya Aura™ Session Manager from Avaya
Aura™ Communication Manager.
e. Enter a description in the Notes field if desired.
2. Originating Locations and Routing Policies Section
a. Click on the Add button and a window will open (not shown).
b. Click on the boxes for the appropriate Originating Locations (see Section 6.3.3),
and Routing Policies (see Section 6.3.7) that pertain to this Dial Pattern.
i. Location 10.64.20/21.0
ii. Routing Policies to Acme_Packet.
c. Click on the Select button and return to the Dial Pattern window.
3. Click the Commit button
4. Repeat steps 1 thru 3 for the remaining Dial Patterns.
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Figure 57: Dial Pattern Details
6.4. Avaya Aura™ Session Manager
To complete the Avaya Aura™ Session Manager configuration, add an Avaya Aura™ Session
Manager instance. To add an Avaya Aura™ Session Manager, select Session Manager on the left
Network Routing Policy menu and click on the New button. The screen shown below is
displayed.
1. General section
a. Enter a name in the SIP Entity Name field (e.g. SM 01).
b. Enter an optional description in the Description field.
c. In the Management Access Point Host Name/IP field enter the IP address of the
management interface of the Avaya Aura™ Session Manager server. (e.g.
10.64.20.30).
2. Security Module section
Note – The SIP Entity IP address is automatically populated with the IP address defined for this
SIP Entity (SM 01) in Section 6.3.4.
a. Enter the Network Mask (e.g. 255.255.255.0)
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b. Enter the Default Gateway (e.g. 10.64.20.1)
c. In the Speed & Duplex drop down menu verify Auto is selected (default).
3. Use all other default parameters.
4. Click the Save button and the completed form will be displayed.
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Figure 58: Completed Session Manager Form
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6.5. Feature Server
In order for Communication Manager to provide configuration and Feature Server support to the
Avaya 9600 Series SIP Telephones when they register to Session Manager, Communication
Manager must be added as an application for Session Manager.
1. Select Applications Entities on the left. Click on New (not shown). Select “CM” Type
and in the displayed page, enter the following fields. Use defaults for the remaining fields:
a. Enter a descriptive name in the Name field.
b. Select CM for Type.
c. In the Node field, Select IP address for Communication Manager SAT access.
Under the Attributes section, enter the following fields, and use defaults for the
remaining fields:
d. Enter the login used for SAT access in the Login field.
e. Enter the password used for SAT access in the Password field.
f. Enter the password used for SAT access in the Confirm Password field.
2. Click the Commit button.
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Figure 59: Application Entities Form
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3. Select Session Manager Application Configuration Applications on the left. Click
on New (not shown). Enter the following fields, and use defaults for the remaining fields:
a. Enter a descriptive name in the Name field.
b. Select the Communication Manager SIP Entity (see Section 6.3.4) for SIP Entity.
4. Click the Commit button.
Figure 60: Application Editor Form
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5. Select Session Manager Application Configuration Application Sequences on the
left. Click on New (not shown).
a. Enter a descriptive name in the Name field.
b. Click on the “+” sign next to the appropriate Available Applications, and the
selected available application will be moved up to the Applications in this Sequence
section.
6. Click the Commit button.
Figure 61: Application Sequence Editor Form
7. Select Communication System Management Telephony on the left.
a. Select the appropriate Element Name.
b. Select Initialize data for selected devices radio button.
c. Click the Now button. This will cause a data synchronization task to start. This may
take some time to complete.
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Figure 62: Synchronize CM Data and Configure Options Form
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6.6. User Management for Adding SIP Telephone Users
SIP users must be added to Session Manager.
1. Select User Management User Management on the left. Then click on New (not
shown) to open the New User Profile page.
a. Enter a First Name and Last Name for the user.
Figure 63: New User Profile Form
2. Click on Identity to expand that section. Enter the following fields, and use defaults for the
remaining fields:
a. Login Name:
<extension>@avaya.com
b. SMGR Login Password:
c. Password:
Password to log into System Manager
d. Confirm Password:
Password to log into System Manager
e. Shared Communication
f. Profile Password:
Password to be entered by the user when logging onto
the telephone
g. Confirm Password:
Password to be entered by the user when logging onto
the telephone
h. Localized Display Name: Name to be used as calling party
i. Endpoint Display Name: Full name of user
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j. Language Preference:
k. Time Zone:
Select the appropriate language preference
Select the appropriate time zone
Figure 64: New User Profile Form – cont’d
3. Click on Communication Profile to expand that section in the above screen. Then click on
Communication Address to expand that section. Enter the following fields and use
defaults for the remaining fields:
a. Type:
Select “sip”
b. SubType:
Select “username”
c. Fully Qualified Address: Enter the extension and select the domain as specified
in Section 6.3.1
d. Click on Add to add the record with the above information.
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Figure 65: New User Profile Form – cont’d
4. Click on Session Manager in the above screen to expand that section. Select the
appropriate Session Manager server for Session Manager Instance. For Origination
Application Sequence and Termination Application Sequence, select the Application
Sequence configured in Section 6.5.
5. Click on Station Profile in the above screen to expand that section. Enter the following
fields and use defaults for the remaining fields:
a. System:
Select the Communication Manager entity.
b. Use Existing Stations:
Check this box.
c. Extension:
Enter the extension.
d. Template:
Select an appropriate template matching the telephone
type.
e. Port:
Click on the Search icon to pick a port (in this case
(“IP”)
6. Click on Commit (not shown).
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Figure 66: New User Profile Form – cont’d
7. Repeat the above procedures to add each SIP user.
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7. Acme Packet 3800 Net-Net Session Director
In the reference configuration, an Acme Packet Session Border Controller (SBC) was used to
provide access to the Metaswitch network.
7.1. Acme Packet Service States
Acme Packet requests and provides service states by sending out and responding to SIP OPTIONS
messages. Acme Packet sends the OPTIONS message with the hop count (SIP Max-Forwards) set
to zero.
•
•
Acme Packet/Avaya Aura™ Session Manager
o Acme Packet sends OPTIONS Avaya Aura™ Session Manager responds with
200 OK
o Avaya Aura™ Session Manager sends OPTIONS Acme Packet responds with
200 OK
Acme Packet/Metaswitch
o Acme Packet sends OPTIONS Metaswitch responds with 200 OK
o Metaswitch sends OPTIONS Acme Packet responds with 200 OK
7.2. Acme Packet Network Interfaces
The physical and network interface provisioning for the “OUTSIDE” (to Metaswitch) and
“INSIDE” (to Avaya CPE) interfaces is described in Sections 7.3.3 and 7.3.4.
7.3. Acme Packet Provisioning
Note – Only the Acme Packet provisioning required for the reference configuration is described in
these Application Notes.
The Acme Packet SBC was configured using the Acme Packet CLI via a serial console port
connection. An IP remote connection to a management port is also supported. The following are
the generic steps for configuring various elements.
1. Log in with the appropriate credentials.
2. Enable the Superuser mode by entering enable command and the appropriate password
(prompt will end with #).
3. In Superuser mode, type configure terminal and press <ENTER>. The prompt will
change to (configure)#.
4. Type the name of the element that will be configured (e.g., session-router).
5. Type the name of the sub-element, if any (e.g., session-agent).
6. Type the name of the parameter followed by its value (e.g., ip-address).
7. Type done.
8. Type exit to return to the previous menu.
9. Repeat steps 4-8 to configure all the elements. When finished, exit from the
configuration mode by typing exit until returned to the Superuser prompt.
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10. Type save-configuration to save the configuration.
11. Type activate-configuration to activate the configuration.
Once the provisioning is complete, the configuration may be verified by entering the show
running-config command.
7.3.1 Acme Packet Management
Initial Acme Packet provisioning is performed via the console serial port (115200,
8/None/1/None). Network management is enabled by provisioning interface “Wancom0”. In the
reference configuration, the management IP address 169.254.1.1 is assigned.
From the configure prompt (steps 1 thru 3 in Section 7.3):
1. Enter bootparam
Note - This command will prompt one line at a time showing the existing value. Enter
the new value next to the existing value. If there is no change to a value, hit the enter
key and the next line will be presented. Be careful not to modify any values other than
those listed below, or the Acme Packet may not recover after a reboot.
Console output will appear as follows:
acmesbc-pri(configure)# bootparam
'.' = clear field; '-' = go to previous field; q = quit
boot device
: wancom0
2. Press Enter at the boot device : wancom0 line, and the next 4 lines until the following
is displayed:
inet on ethernet (e)
:
3. Enter the IP address and mask (in hex) to be used for network management (e.g.
169.254.1.1:ffffff00) and press Enter 3 more times until the following is displayed:
gateway inet (g)
:
4. Enter the management network gateway IP address (e.g. 169.254.1.1) and press Enter.
5. Continue to press Enter until returned to the “configure” prompt. After the last
bootparam line, the following message is displayed:
NOTE: These changed parameters will not go into effect until reboot. Also, be aware
that some boot parameters may also be changed through PHY and Network Interface
Configurations.
6. At the “configure” prompt enter exit
7. Reboot the Acme Packet by entering reboot at the Superuser “#” prompt.
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7.3.2 Local Policies
Allows any SIP requests from the INSIDE realm to be routed to the SERV_PROVIDER Session
Agent Group in the OUTSIDE realm (and vice-versa).
7.3.2.1 INSIDE to OUTSIDE
From the configure prompt (steps 1 thru 3 in Section 5.3):
1. Configure session-router local-policy to create a local-policy for the INSIDE realm
with the following settings:
a. from-address *
b. to-address *
c. source-realm INSIDE
d. state enabled
e. policy-attributes
i. next-hop SAG:SERV_PROVIDER
ii. realm OUTSIDE
iii. start-time 0000
iv. end-time 2400
v. days-of-week U-S
vi. app-protocol SIP
vii. state enabled
7.3.2.2 OUTSIDE to INSIDE
1. Configure session-router local-policy to create a local-policy for the OUTSIDE
realm with the following settings:
a. from-address *
b. to-address *
c. source-realm OUTSIDE
d. state enabled
e. policy-attributes
i. next-hop SAG:ENTERPRISE
ii. realm INSIDE
iii. start-time 0000
iv. end-time 2400
v. days-of-week U-S
vi. app-protocol SIP
vii. state enabled
7.3.3 Network Interfaces
This Section defines the network interfaces to the private (Avaya CPE) and public (Metaswitch) IP
networks.
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7.3.3.1 Public Network Interface
1. Configure system network-interface to create a network-interface to the public
(Internet/Metaswtich) side of the Acme Packet with the following settings:
a. name Public
b. ip-address 205.xxx.xxx.106
c. netmask 255.255.255.128
d. gateway 205.xxx.xxx.1
7.3.3.2 Private Network Interface
1. Configure system network-interface to create a network-interface to the private
(Avaya CPE) side of the Acme Packet with the following settings:
a. name Private
b. ip-address 10.64.20.106
c. netmask 255.255.255.0
d. gateway 10.64.20.1
7.3.4 Physical Interfaces
This Section defines the physical interfaces to the private (Avaya CPE) and public (Metaswitch)
networks.
7.3.4.1 Public Physical Interface
1. Configure system phy-interface to create a network-interface to the public
(Internet/Metaswitch) side of the Acme Packet with the following settings:
a. name Public
b. operation-type media
c. port 0
d. slot 0
e. virtual-mac 00:08:25:A0:E2:28
i. Virtual MAC addresses are assigned based on the MAC address assigned to
the Acme. This MAC address is found by entering the command show
prom-info mainboard (e.g. 00:08:25:A0:E2:20). To define a virtual MAC
address, replace the last digit with 8 thru f.
f. duplex-mode FULL
g. speed 100
7.3.4.2 Private Physical Interface
1. Configure system phy-interface to create a phy-interface to the private (Avaya
CPE) side of the Acme Packet with the following settings:
a. name Private
b. operation-type media
c. port 0
d. slot 1
e. virtual-mac 00:08:25:A0:E2:2e
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f. duplex-mode FULL
g. speed 100
7.3.5 Realms
Realms are used as a basis for determining egress and ingress associations between physical and
network interfaces as well as applying header manipulation such as NAT.
7.3.5.1 Outside Realm
1. Configure media-manager realm-config to create a realm for the outside network
with the following settings:
a. identifier OUTSIDE
b. addr-prefix 0.0.0.0
c. network-interfaces Public:0
d. out-manipulationid NAT_IP
e. mm-in-realm enabled
f. mm-in-network enabled
g. mm-same-ip enabled
h. mm-in-system enabled
i. access-control-trust-level medium
j. invalid-signal-threshold 1
k. maximum-signal-threshold 1
l. untrusted-signal-threshold 1
7.3.5.2 Inside Realm
1. Configure media-manager realm-config to create a realm for the inside network
with the following settings:
a. identifier INSIDE
b. addr-prefix 0.0.0.0
c. network-interfaces Private:0
d. out-manipulationid NAT_IP
e. mm-in-realm enabled
f. mm-in-network enabled
g. mm-same-ip enabled
h. mm-in-system enabled
i. access-control-trust-level high
j. invalid-signal-threshold 0
k. maximum-signal-threshold 0
l. untrusted-signal-threshold 0
7.3.6 Steering-Pools
Steering pools define sets of ports that are used for steering media flows thru the Acme.
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7.3.6.1 Outside Steering-Pool
1. Configure media-manager steering-pool to create a steering-pool for the outside
network with the following settings:
a. ip-address 205.xxx.xxx.106
b. start-port 49152
c. end-port 65535
d. realm-id OUTSIDE
7.3.6.2 Inside Steering-Pool
1. Configure media-manager steering-pool to create a steering-pool for the inside
network with the following settings:
a. ip-address 10.64.20.106
b. start-port 49152
c. end-port 65535
d. realm-id INSIDE
7.3.7 Session-Agents
A session-agent defines an internal “next hop” signaling entity for the SIP traffic. A realm is
associated with a session-agent to identify sessions coming from or going to the session-agent. A
session-agent is defined for the Metaswitch service node (outside) and for the Avaya Aura™
Session Manager (inside).
7.3.7.1 Outside Session-Agent
1. Configure session-router session-agent to create a session-agent for the outside
network with the following settings:
a. hostname 208.xxx.xxx.135
b. ip-address 208.xxx.xxx.135
c. port 5060
d. state enabled
e. app-protocol SIP
f. transport-method UDP
g. realm-id OUTSIDE
h. description To_Metaswitch
i. ping-method Options;hops=0
j. ping-interval 60
k. ping-send-mode keep-alive
7.3.7.2 Inside Session-Agent
1. Configure session-router session-agent to create a session-agent for the inside
network with the following settings:
a. hostname 10.64.20.31
b. ip-address 10.64.20.31
c. port 5060
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d.
e.
f.
g.
h.
i.
j.
k.
l.
m.
state enabled
app-protocol SIP
transport-method staticTCP
realm-id INSIDE
description To_Session Manager
ping-method Options;hops=0
ping-interval 60
ping-send-mode keep-alive
tcp-keepalive enabled
tcp-reconn-interval 10
7.3.8 Session Groups
Session-groups (SAG) define single or multiple destinations that are referenced in provisioning
session-agents.
7.3.8.1 Metaswitch Session-group
1. Configure session-router session-group to create a session-group for the
Metaswitch network with the following settings:
a. groupname SERV_PROVIDER
b. state enabled
c. app-protocol SIP
d. strategy Hunt
e. dest 208.xxx.xxx.135
7.3.8.2 Avaya CPE Session-group
1. Configure session-router session-group to create a session-group for the Avaya
CPE network with the following settings:
a. groupname ENTERPRISE
b. state enabled
c. app-protocol SIP
d. strategy Hunt
e. dest 10.64.20.31
7.3.9 SIP Configuration
This command sets the values for the Acme Packet SIP operating parameters. The home-realm
defines the SIP daemon location, and the egress-realm is the realm that will be used to send a
request if a realm is not specified elsewhere.
1. Configure session-router sip-config with the following settings:
a. state enabled
b. operation-mode dialog
c. home-realm-id INSIDE
d. egress-realm-id OUTSIDE
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7.3.10 SIP Interfaces
The SIP interface defines the signaling interface (IP address and port) to which the Acme Packet
sends and receives SIP messages.
7.3.10.1 Outside SIP- interface
1. Configure session-router sip-interface to create a sip-interface for the outside
network with the following settings:
a. state enabled
b. realm-id OUTSIDE
c. sip-port i. address 205.xxx.xxx.106
ii. port 5060
iii. transport-protocol UDP
7.3.10.2 Inside SIP- interface
1. Configure session-router sip-interface to create a sip-interface for the inside
network with the following settings:
a. state enabled
b. realm-id INSIDE
c. sip-port
i. address 10.64.20.106
ii. port 5060
iii. transport-protocol TCP
7.3.11 SIP Manipulation
SIP manipulation specifies rules for manipulating the contents of specified SIP headers. In the
reference configuration the following header manipulations are performed:
•
•
•
•
•
NAT IP addresses in the From header of SIP requests.
NAT IP addresses in the To header of SIP requests.
NAT IP addresses in the Remote-Party-ID header of SIP requests.
NAT IP addresses in the History-Info header of SIP requests.
NAT IP addresses in the Alert-Info header of SIP requests. This is different from other
rules because it will NAT CID (caller ID) URIs in addition to SIP URIs.
1. Configure session-router sip-manipulation with the following settings :
a. name NAT_IP
b. description Topology hiding for SIP headers
2. Proceed to the following sections.
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7.3.11.1 From Header
1. Configure session-router sip-manipulation header-rule with the following
settings:
a. name manipFrom
b. action manipulate
c. comparison-type case-sensitive
d. msg-type request
e. element-rule
i. name FROM
ii. type uri-host
iii. action replace
iv. match-val-type ip
v. comparison-type case-sensitive
vi. new-value $LOCAL_IP
7.3.11.2 To Header
1. Configure session-router sip-manipulation header-rule with the following
settings:
a. name manipTo
b. action manipulate
c. comparison-type case-sensitive
d. msg-type request
e. element-rule
i. name TO
ii. type uri-host
iii. action replace
iv. match-val-type ip
v. comparison-type case-sensitive
vi. new-value $REMOTE_IP
7.3.11.3 Remote Party ID Header
1. Configure session-router sip-manipulation header-rule with the following
settings:
a. name manipRpid
b. header-name Remote-Party-ID
c. action manipulate
d. comparison-type case-sensitive
e. msg-type request
f. element-rule
i. name RPID
ii. type uri-host
iii. action replace
iv. match-val-type ip
v. comparison-type case-sensitive
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vi. new-value $LOCAL_IP
7.3.11.4 History Info Header
1. Configure session-router sip-manipulation header-rule with the following
settings:
a. name manipHistInfo
b. header-name History-Info
c. action manipulate
d. comparison-type case-sensitive
e. msg-type request
f. element-rule
i. name HISTORYINFO
ii. type uri-host
iii. action replace
iv. match-val-type ip
v. comparison-type case-sensitive
vi. new-value $REMOTE_IP
7.3.11.5 Alert-info Header
1. Configure session-router sip-manipulation header-rule with the following
settings:
a. name storeAlertInfo
b. header-name Alert-Info
c. action store
d. comparison-type pattern-rule
e. match-value (.+@) ([0-9.]+) (.+)
f. msg-type request
2. Configure session-router sip-manipulation header-rule with the following
settings:
a. name manipAlertInfo
b. header-name Alert-Info
c. action manipulate
d. comparison-type boolean
e. match-value $storeAlertInfo
f. msg-type request
g. new-value $storeAlertInfo.$1+$REMOTE_IP+$storeAlertInfo.$3
7.3.12 Other Acme Packet provisioning
7.3.12.1 Access-control
This is a static Access Control List that is used to limit SIP access to only known devices.
1. Configure session-router access-control with the following settings:
a. realm-id OUTSIDE
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b.
c.
d.
e.
source-address 208.xxx.xxx.135:5060
application-protocol SIP
transport-protocol UDP
access permit
7.3.12.2 Media-Manager
Verify that the media-manager process is enabled.
1. Navigate to media-manager media-manager
2. Enter select show Verify that the media-manager state is enabled. If not,
configure the following settings:
a. state enabled
7.3.12.3 System-config
In the system-config, specify a hostname and the default gateway of the management interface.
1. Configure system system-config with the following settings:
a. hostname acmesbc
b. default-gateway 10.64.20.1
8. Metaswitch Configuration
During the test effort, the Metaswitch network was protected by a pair of Acme Packet Net-Net SD
3820 session border controllers. The session border controllers are not required as part of the
solution. For brevity, only the configuration of the MetaSphere CFS is discussed below. If a
session border controller is used between the MetaSphere CFS solution and the Avaya solution,
contact a Metaswitch Networks support representative for additional configuration details.
8.1. Media Gateway Model
A truncated text dump of the Remote Media Gateway Model used for the testing is shown below.
For an importable version, contact a Metaswitch customer service representative.
begin MediaGatewayModel MediaGatewayModel.176 // Remote Media Gateway Model "Avaya
CM/SM"
Category
SIP
ModelName
Avaya CM/SM
ControlProtocol
SIP
DefaultModel
False
SupportedHighBandwidthMediaFormats
{G.711 u-law,G.711 A-law}
SupportedLowBandwidthMediaFormats
{G.729 AB}
PreferredLowBandwidthMediaFormats
{G.729 AB}
AdvancedVoiceCodecsPermitted
Any codecs
VideoCodecsPermitted
Any codecs
PacketizationInterval
0
SilenceSuppressionAllowed
False
MaximumSimultaneousTransactionsOutstanding
100
DigitOverhangTime
250
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FixBitsMGCPMeGaCoSIP
DynamicFixBitsMGCPMeGaCoSIP
FixBitsSIP
FixBitsSIP2
ReferenceCount
UpToDateCount
ExportHeading
StatusHeading
RequestedStatus
end //MediaGatewayModel
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{Cannot be hub,Simple contexts,Cannot
control endpoint connectivity,Cannot
move contexts,Connections always
receive,Cannot report detection of
call-type discrimination tones,T.38
supported}
{}
{Supports SDP connectivity
requests,Supports receiving INVITEs
with no SDP,Supports receiving SIP
Reason header over tandem trunk
calls}
{}
1
1
Export
Status
Enabled
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8.2. Configured SIP Binding
The connection to the Avaya solution is modeled as a configured SIP binding. During compliance
testing, the configured SIP binding was configured as follows.
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8.3. PBX object configuration
The Avaya solution is modeled in MetaView as a PBX. The settings used during testing are shown
below.
8.3.1 PBX Object
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8.3.2 PBX Line Object
8.3.3 DID objects
Two DID ranges were configured during compliance testing due to the setup in Metaswitch’s test
lab. Two DID ranges are not required.
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9. General Test Approach and Test Results
This section describes the interoperability compliance testing used to verify SIP trunking
interoperability between Metaswitch and the Avaya CPE. This section covers the general test
approach and the test results.
The Avaya CPE was connected using SIP trunking to the Metaswitch network. The general test
approach included the following:
•
Inbound Calls – Verify that calls placed from a PSTN telephone to a DID number are
properly routed via the SIP trunk to the expected extension on Communication Manager.
Verify the talk-path exists in both directions, that calls remain stable for several minutes
and disconnect properly.
•
Outbound Calls – Verify that calls placed to a PSTN telephone are properly routed via the
SIP trunk. Verify that the talk-path exists in both directions and that calls remain stable and
disconnect properly.
•
Inbound DTMF Digit Navigation – Verify inbound DID calls can properly navigate voice
mail menus.
2.
Outbound DTMF Digit Navigation – Verify outbound calls can properly navigate a voice
mail or interactive response system reached via a PSTN number.
Interoperability testing of the reference configuration was completed with successful results.
The following observations were noted:
1. The Metaswitch test lab did not support x11 calls (e.g 411, 911, international, etc.).
2. Due to limitations of the Metaswitch test lab configuration, the caller-id did not always
display the proper calling party number for calls to and from the PSTN (rather, an
administered 10 digit number was display).
10. Verification Steps
This Section provides the verification steps that may be performed to verify basic operation of the
Avaya Aura™ SIP trunk solution with Metaswitch.
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10.1. Verify Avaya Aura™ Communication Manager 5.2
Verify the status of the SIP trunk group by using the status trunk n command, where “n” is the
administered trunk group number. Verify that all trunks are in the “in-service/idle” state as shown
below.
status trunk 9
TRUNK GROUP STATUS
Member
Port
Service State
Mtce Connected Ports
Busy
0009/001
0009/002
0009/003
0009/004
0009/005
0009/006
0009/007
0009/008
0009/009
0009/010
T00334
T00335
T00336
T00337
T00338
T00339
T00340
T00341
T00342
T00343
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
in-service/idle
no
no
no
no
no
no
no
no
no
no
Figure 67: Status Trunk
Verify the status of the SIP signaling groups by using the status signaling-group n command,
where “n” is the administered signaling group number. Verify the signaling group is “in-service”
as indicated in the Group State field shown below.
status signaling-group 9
STATUS SIGNALING GROUP
Group ID:
Group Type:
Signaling Type:
Group State:
9
sip
facility associated signaling
in-service
Active NCA-TSC Count: 0
Active CA-TSC Count: 0
Figure 68: Status Signaling Group
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Make a call between an Avaya Aura™ Communication Manager H.323 station and the PSTN.
Verify the status of connected SIP trunk by using the status trunk x/y command, where “x” is the
number of the SIP trunk group, and “y” is the active member number of a connected trunk. Verify
on Page 1 that the Service State is “in-service/active”. On Page 2, verify that the IP addresses of
the procr or C-LAN and Avaya Aura™ Session Manager are shown in the Signaling section. In
addition, the Audio section shows the G.711 codec and the IP address of the Avaya H.323
endpoint and the Acme Packet SBC. The Audio Connection Type displays “ip-direct”,
indicating direct media between the two endpoints.
status trunk 9/7
Page
1 of
3
TRUNK STATUS
Service State: in-service/active
Maintenance Busy? no
Trunk Group/Member: 0009/007
Port: T00340
Signaling Group ID: 9
IGAR Connection? no
Connected Ports: S00027
Figure 69: Status Trunk – Active Call – Page 1
status trunk 9/7
Page
2 of
3
CALL CONTROL SIGNALING
Near-end Signaling Loc: 01A0017
Signaling
IP Address
Near-end: 10.64.21.41
Far-end: 10.64.20.31
H.245 Near:
H.245 Far:
H.245 Signaling Loc:
Port
: 5061
: 5061
H.245 Tunneled in Q.931? no
Audio Connection Type: ip-direct
Near-end Audio Loc:
Audio
IP Address
Near-end: 10.64.21.71
Far-end: 10.64.20.106
Authentication Type: None
Codec Type: G.711MU
Port
: 2662
: 50248
Video Near:
Video Far:
Video Port:
Video Near-end Codec:
Video Far-end Codec:
Figure 70: Status Trunk – Active Call – Page 2
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10.2. Verify Avaya Aura™ Session Manager
Monitoring of Avaya Aura™ Session Manager is performed via Avaya Aura™ System Manager.
10.2.1
Verify SIP Entity Link Status
Expand the Session Manager menu and click SIP Monitoring. Verify that none of the links to the
defined SIP entities are down (as indicated by 0/4 in the figure below), indicating that they are all
reachable for call routing.
Figure 71: SIP Entity Link Monitoring - Summary
Selecting a monitored SIP Entity from the list will display its status (e.g. S8300_G450_AE).
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Figure 72: SIP Entity Link Connection Status
10.2.2
Verify System State
Expand the Session Manager menu and click System State Administration. Verify that the
Management State is Management Enabled and the Service State is Accept New Service.
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Figure 73: System State
10.3. Verification Call Scenarios
Verification scenarios for the configuration described in these Application Notes included:
• Inbound and outbound basic voice calls between various telephones on the Avaya Aura™
Communication Manager and PSTN can be made in both directions.
o Avaya One-X Communicator (H.323 Softphone), as well as traditional analog,
digital, and SIP phones.
• Inbound and outbound Fax calls between Avaya Aura™ Communication Manager and
PSTN can be made in both directions.
• Direct IP-to-IP Media (also known as “Shuffling”) when applicable.
• DTMF Tone Support.
• Additional PSTN numbering plans.
• Supplementary calling features were verified. The supplementary calling features verified
are:
o Hold, Call transfer, Conference.
o Voicemail Coverage and Retrieval.
o Call Forwarding.
o Call Coverage.
o Extend Call.
o EC500 (call forking).
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10.4. Conclusion
As illustrated in these Application Notes, Avaya Aura™ Communication Manager 5.2.1, Avaya
Aura™ Session Manager 5.2.1.1, and Acme Packet Session Border Controller 6.1.0 can be
configured to interoperate successfully with Metaswitch MetaSphere CFS. This solution provides
users of Avaya Aura™ Communication Manager the ability to support inbound and outbound as
well as on-net and off-net calling over a SIP trunk.
11. References
11.1. Avaya
The following Avaya product documentation is available at http://support.avaya.com.
[1] SIP Support in Avaya Aura™ Communication Manager Running on Avaya S8xxx Servers,
Doc ID 555-245-206, May 2009.
[2] Administering Avaya Aura™ Communication Manager, Doc ID 03-300509, May 2009.
[3] Avaya Aura™ Session Manager Overview, Doc ID 03-603323, Issue 2, Release 5.2, March
2010.
[4] Installing Avaya Aura™ Session Manager, Doc ID 03-603473, Issue 1.3, Release 5.2,
January 2010.
[5] Administering Avaya Aura™ Session Manager, Doc ID 03-603324, Issue 2, Release 5.2,
November 2009.
[6] Administering Avaya Aura™ Communication Manager as a Feature Server, Doc ID 03603479, Issue 1.3, Release 5.2, March 2010
[7] Maintaining and Troubleshooting Avaya Aura™ Session Manager, Doc ID 03-603325.
[8] Feature Description and Implementation for Avaya Communication Manager, Doc ID555245-205, Issue 7, Release 5.2, May 2009
11.2. Metaswitch
Metaswitch product documentation is available at http://www.metaswitch.com/support/.
11.3. Acme Packet
The following Acme Packet product documentation is available at:
https://support.acmepacket.com/
[9] Net-Net® 4000, ACLI Reference Guide, Release Version S-C6.1.0
[10] Net-Net® 4000 ACLI, Configuration Guide, Release Version S-C6.1.0
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©2010 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks
are the property of their respective owners. The information provided in these Application
Notes is subject to change without notice. The configurations, technical data, and
recommendations provided in these Application Notes are believed to be accurate and
dependable, but are presented without express or implied warranty. Users are responsible for
their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya
DevConnect Program at [email protected].
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