Application Notes for Configuring SingTel Meg@POP SIP Trunking

Application Notes for Configuring SingTel Meg@POP SIP Trunking
Avaya Solution & Interoperability Test Lab
Application Notes for Configuring SingTel Meg@POP SIP
Trunking Service with Avaya Aura® Communication
Manager 6.0.1, Avaya Aura® Session Manager 6.1, and
Acme Packet 4250 Net-Net Session Border Controller
– Issue 1.0
Abstract
These Application Notes describe the steps to configure Session Initiation Protocol (SIP)
Trunking between SingTel Meg@POP SIP Trunking Service and an Avaya SIP-enabled
enterprise solution. The Avaya solution consists of Avaya Aura® Session Manager, Avaya
Aura® Communication Manager, Acme Packet 4250 Net-Net Session Border Controller and
various Avaya endpoints.
Information in these Application Notes has been obtained through DevConnect compliance
testing and additional technical discussions. Testing was conducted via the DevConnect
Program at the Avaya Solution and Interoperability Test Lab.
JC; Reviewed:
SPOC 11/8/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
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Table of Contents
1.
2.
Introduction ............................................................................................................................. 3
General Test Approach and Test Results ................................................................................ 3
2.1. Interoperability Compliance Testing ................................................................................ 3
2.2. Test Results ...................................................................................................................... 4
2.3. Support ............................................................................................................................. 4
3. Reference Configuration.......................................................................................................... 5
4. Equipment and Software Validated ......................................................................................... 7
5. Configure Avaya Aura® Communication Manager................................................................ 8
5.1. Licensing and Capacity .................................................................................................... 8
5.2. System Features................................................................................................................ 9
5.3. IP Node Names............................................................................................................... 10
5.4. Codecs ............................................................................................................................ 10
5.5. IP Network Region ......................................................................................................... 11
5.6. Signaling Group ............................................................................................................. 13
5.7. Trunk Group ................................................................................................................... 15
5.8. Calling Party Information............................................................................................... 17
5.9. Outbound Call Routing .................................................................................................. 18
5.10.
Inbound Call Routing ................................................................................................. 20
6. Configure Avaya Aura® Session Manager ........................................................................... 20
6.1. Avaya Aura® System Manager Login and Navigation ................................................. 21
6.2. Specify SIP Domain ....................................................................................................... 23
6.3. Configure Location ........................................................................................................ 24
6.4. Add Adaptation Module ................................................................................................. 25
6.5. Add SIP Entities ............................................................................................................. 26
6.6. Add Entity Links ............................................................................................................ 30
6.7. Add Routing Policies ..................................................................................................... 32
6.8. Add Dial Patterns ........................................................................................................... 33
6.9. Add/View Session Manager ........................................................................................... 36
7. Configure Acme Packet 4250 Net-Net Session Border Controller ....................................... 37
8. SingTel Meg@POP SIP Trunking Service Configuration .................................................... 38
9. Verification Steps .................................................................................................................. 38
10. Conclusion ......................................................................................................................... 39
11. Additional References ........................................................................................................ 39
12. Appendix A: Acme Packet 4250 Net-Net Session Border Controller Configuration File 40
JC; Reviewed:
SPOC 11/8/2011
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1. Introduction
These Application Notes describe the steps to configure Session Initiation Protocol (SIP)
Trunking between SingTel Meg@POP SIP Trunking Service and an Avaya SIP-enabled
enterprise solution. The Avaya solution consists of Avaya Aura® Session Manager, Avaya
Aura® Communication Manager, Acme Packet 4250 Net-Net Session Border Controller (SBC)
and various Avaya endpoints.
SingTel Meg@POP SIP Trunking Service provides businesses with multiple location seamless
access to Public Switched Telephone Network (PSTN). By converging voice and data services
onto a single Meg@POP network, customers enjoy cost savings by simplifying their network
infrastructure, and optimizing the existing network.
2. General Test Approach and Test Results
The general test approach was to connect a simulated enterprise site to the SingTel Meg@POP
SIP Trunking Service and exercise the features and functionality listed in Section 2.1. The
simulated enterprise site was comprised of Communication Manager, Session Manager and the
Acme Packet 4250 Net-Net SBC. Testing was done in the SingTel lab environment that
simulated the actual SingTel Meg@POP SIP Trunking Service. Acme Packet 4250 Net-Net SBC
was also provided and provisioned by Acme Packet engineers for the testing.
2.1. Interoperability Compliance Testing
To verify SIP trunking interoperability, the following features and functionality were covered
during the interoperability compliance test:












Response to SIP OPTIONS queries.
Incoming PSTN calls to various phone types including H.323, SIP, digital, and analog
telephones at the enterprise. All inbound PSTN calls were routed to the enterprise across
the SIP trunk from the service provider.
Outgoing PSTN calls from various phone types including H.323, SIP, digital, and analog
telephones at the enterprise. All outbound PSTN calls were routed from the enterprise
across the SIP trunk to the service provider.
Inbound and outbound PSTN calls to/from Avaya one-X® Communicator (soft client).
Avaya one-X Communicator supports two modes (Road Warrior and Telecommuter).
Both supported modes were tested. Both H.323 and SIP protocols were tested.
Codecs G.711A and G.729A were tested.
DTMF transmission using RFC 2833.
Caller ID presentation and Caller ID restriction.
Response to incomplete call attempts and trunk errors.
Voicemail navigation for inbound and outbound calls.
User features such as hold and resume, internal call forwarding, transfer, and conference.
Off-net call forwarding and mobility (extension to cellular).
Incoming PSTN calls to a Vector Directory Number (VDN) and delivered to agents.
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Items not supported or not tested included the following:



Long distance, international, outbound toll-free and operator assisted calls were not tested
due to limitations in the test environment.
T.38 Fax is not supported.
Use of the REFER method and a 302 redirected response were not tested.
For the compliance test, the enterprise sent the dialed digits in non-E.164 format (e.g. 68591234,
00113035381234) in the destination headers (e.g., “Request-URI” and “To”) and sent 10 digits
in E.164 format (e.g. +6568596789) in the source headers (e.g., From, Contact, and P-AssertedIdentity (PAI)). SingTel sent 10 digits in E.164 format in the destination headers and 8 digits in
non-E.164 format in the source headers.
2.2. Test Results
Interoperability testing of SingTel Meg@POP SIP Trunking Service was completed with
successful results for all test cases with the exception of the observations/limitations described
below.



G.729A codec negotiation with Linksys SPA941 phone fails. SingTel Meg@POP
network contains other third-party SIP endpoints. Linksys phone sends “G729a” in the
SDP description, instead of “G729” as defined in RFC4856. As such, Communication
Manager does not accept the call.
G.729A definition in the SDP description not consistent for different third-party SIP
endpoints in SingTel network. As such, G.729A codec negotiation with some PSTN
endpoints may fail. Work around: The IP Codec listed in Section 5.4 has been tested to
work successfully with all the SIP endpoints used in the testing.
Incoming call when all trunks are busy. Communication Manager sends “404 Not
Found”, which SingTel interprets as “Number not in service”. SingTel expects to receive
“486 Busy Here”. Work around: Define the maximum number of trunk members in
Communication Manager and allow SingTel to monitor the SIP trunk usage and plays
network announcement/busytone to caller.
2.3. Support
For technical support on SingTel SIP Trunking Service on the SingTel Meg@POP IP VPN
Network, contact the SingTel Account Manager assigned by SingTel or dial 1800-763-1111 for
general enquiries.
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SPOC 11/8/2011
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3. Reference Configuration
Figure 1 illustrates a sample Avaya SIP-enabled enterprise solution connected to SingTel
Meg@POP SIP Trunking Service. This is the configuration used for compliance testing.
The Avaya components used to create the simulated customer site included:
 Avaya S8800 Server running Avaya Aura® Solution for Midsize Enterprise
o Avaya Aura® Session Manager
o Avaya Aura® System Manager
o Avaya Aura® Communication Manager
o Avaya Aura® Communication Manager Messaging
 Avaya G430 Media Gateway
 Avaya 9600-Series IP telephones (H.323 and SIP)
 Avaya 1600-Series IP telephones (H.323)
 Avaya 1400-Series Digital telephones
 Avaya one-X® Communicator (H.323 and SIP)
 Avaya analog telephone
Located at the edge of the enterprise is the Acme Packet Net-Net 4250 SBC. This was provided
and provisioned by Acme Packet engineers for the testing. It has a public side that connects to
the external network and a private side that connects to the enterprise network. All SIP and RTP
traffic entering or leaving the enterprise flows through the SBC. In this way, the SBC can protect
the enterprise against any SIP-based attacks. The SBC provides network address translation at
both the IP and SIP layers. For security reasons, any actual public IP addresses used in the
configuration have been replaced with private IP addresses. Similarly, any references to real
routable PSTN numbers have also been changed to numbers that can not be routed by the PSTN.
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Figure 1: SingTel Meg@POP SIP Trunking Service Test Configuration
A separate trunk group was created between Communication Manager and Session Manager to
carry the service provider traffic. This was done so that any trunk group or codec settings
required by the service provider could be applied only to this trunk group without affecting other
enterprise SIP traffic. In addition, this trunk carried both inbound and outbound traffic.
For inbound calls, the calls flow from the service provider to the SBC, then to Session Manager.
Session Manager uses the configured dial patterns (or regular expressions) and routing policies
to determine the recipient (in this case, Communication Manager) and on which link to send the
call. Once the call arrives at Communication Manager, further incoming call treatment, such as
incoming digit translations and class of service restrictions may be performed.
Outbound calls to the PSTN are first processed by Communication Manager and may be subject
to outbound features such as automatic route selection, digit manipulation and class of service
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restrictions. Once Communication Manager selects the proper SIP trunk, the call is routed to
Session Manager. Session Manager once again uses the configured dial patterns (or regular
expressions) to determine the route to the SBC. From the SBC, the call is sent to SingTel
Meg@POP SIP Trunking Service.
4. Equipment and Software Validated
The following equipment and software were used for the sample configuration provided:
Avaya IP Telephony Solution Components
Component
Release
Avaya S8800 Server running Avaya Aura® Solution for
Midsize Enterprise
- Avaya Aura® Session Manager
6.1 Service Pack 2
- Avaya Aura® System Manager
6.1 Service Pack 2
- Avaya Aura® Communication Manager
6.0.1 Service Pack 3
- Avaya Aura® Communication Manager Messaging
6.0.1 Service Pack 1
Avaya G430 Media Gateway
31.19.2
- MM711AP Analog MM
HW31 FW095
- MM712AP DCP MM
HW07 FW011
Avaya 1608 IP Telephone (H.323)
1.300B
Avaya 9640G IP Telephone (H.323)
3.1 Service Pack 2
Avaya 9641G IP Telephone (SIP)
6.1 Service Pack 3
Avaya one-X® Communicator (H.323 and SIP)
6.1
Avaya 1408 Digital Telephone
n/a
Avaya 6210 Analog Telephone
n/a
Acme Packet 4250 Net-Net Session Border Controller
6.1.0
SingTel Meg@POP SIP Trunking Service Solution Components
Component
Release
Acme Packet Session Border Controller
Not provided
BroadSoft BroadWorks Softswitch
Cisco Gateway
R17
Not provided
Table 1: Equipment and Software Tested
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5. Configure Avaya Aura® Communication Manager
This section describes the procedure for configuring Communication Manager for SingTel
Meg@POP SIP Trunking Service. A SIP trunk is established between Communication Manager
and Session Manager for use by signaling traffic to and from SingTel. It is assumed the general
installation of Communication Manager, Avaya G430 Media Gateway and Session Manager has
been previously completed and thus is not discussed here.
The Communication Manager configuration was performed using the System Access Terminal
(SAT). Some screens in this section have been abridged and highlighted for brevity and clarity in
presentation. Note that the IP addresses and phone numbers shown throughout these Application
Notes have been edited so that the actual public IP addresses of the network elements and public
PSTN numbers are not revealed.
5.1. Licensing and Capacity
Use the display system-parameters customer-options command to verify that the Maximum
Administered SIP Trunks value on Page 2 is sufficient to support the desired number of
simultaneous SIP calls across all SIP trunks at the enterprise including any trunks to the service
provider. The example shows that 12000 SIP trunks are available and 240 are in use. The license
file installed on the system controls the maximum values for these attributes. If a required
feature is not enabled or there is insufficient capacity, contact an authorized Avaya sales
representative to add additional capacity.
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 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:
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12000
18000
12000
18000
128
100
18000
250
12000
12000
522
10
250
128
128
250
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USED
0
1
0
0
0
0
0
1
240
0
0
0
1
0
0
0
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5.2. System Features
Use the change system-parameters feature command to set the Trunk-to-Trunk Transfer
field to all to allow incoming calls from the PSTN to be transferred to another PSTN endpoint.
If for security reasons, incoming calls should not be allowed to transfer back to the PSTN then
leave the field set to none.
change system-parameters features
Page
FEATURE-RELATED SYSTEM PARAMETERS
Self Station Display Enabled? y
Trunk-to-Trunk Transfer: all
Automatic Callback with Called Party Queuing? n
Automatic Callback - No Answer Timeout Interval (rings): 3
Call Park Timeout Interval (minutes): 10
Off-Premises Tone Detect Timeout Interval (seconds): 20
AAR/ARS Dial Tone Required? y
1 of
19
On Page 9, verify that a text string has been defined to replace the Calling Party Number (CPN)
for restricted or unavailable calls. This text string is entered in the two fields highlighted below.
The compliance test used the value of anonymous for both.
change system-parameters features
FEATURE-RELATED SYSTEM PARAMETERS
Page
9 of
19
CPN/ANI/ICLID PARAMETERS
CPN/ANI/ICLID Replacement for Restricted Calls: anonymous
CPN/ANI/ICLID Replacement for Unavailable Calls: anonymous
DISPLAY TEXT
Identity When Bridging: principal
User Guidance Display? n
Extension only label for Team button on 96xx H.323 terminals? n
INTERNATIONAL CALL ROUTING PARAMETERS
Local Country Code: 65
International Access Code: 011
ENBLOC DIALING PARAMETERS
Enable Enbloc Dialing without ARS FAC? n
CALLER ID ON CALL WAITING PARAMETERS
Caller ID on Call Waiting Delay Timer (msec): 200
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5.3. IP Node Names
Use the change node-names ip command to verify that node names have been previously
defined for the IP addresses of the Avaya S8800 Server running Communication Manager
(procr) and for Session Manager (SM). These node names will be needed for defining the
service provider signaling group in Section 5.6.
change node-names ip
Page
1 of
2
IP NODE NAMES
Name
SM
default
procr
procr6
IP Address
10.1.40.24
0.0.0.0
10.1.40.10
::
5.4. Codecs
Use the change ip-codec-set command to define a list of codecs to use for calls between the
enterprise and the service provider. For the SingTel SIP Trunking Service, the preferred codec is
G.729A. However, due to a difference in the way Avaya handles the G.729 MIME Type in the
Session Description Protocol (SDP) parameters, Avaya recommends that the G.729AB codec is
listed before the G.729A codec. To use these codecs, enter G.729AB, G.729A, G.711A and
G.711MU in the Audio Codec column of the table in the order of preference. Default values can
be used for all other fields.
change ip-codec-set 2
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2
IP Codec Set
Codec Set: 1
1:
2:
3:
4:
Audio
Codec
G.729AB
G.729A
G.711A
G.711MU
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Silence
Suppression
n
n
n
n
Frames
Per Pkt
2
2
2
2
Packet
Size(ms)
20
20
20
20
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On Page 2, set the Fax Mode to off since T.38 fax is not supported.
change ip-codec-set 2
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IP Codec Set
Allow Direct-IP Multimedia? y
Maximum Call Rate for Direct-IP Multimedia: 2048:Kbits
Maximum Call Rate for Priority Direct-IP Multimedia: 2048:Kbits
FAX
Modem
TDD/TTY
Clear-channel
Mode
off
off
off
n
Redundancy
0
0
0
0
5.5. IP Network Region
Create a separate IP network region for the service provider trunk. This allows for separate codec
or quality of service settings to be used (if necessary) for calls between the enterprise and the
service provider versus calls within the enterprise or elsewhere. For the compliance test, IPnetwork-region 2 was chosen for the service provider trunk. Use the change ip-network-region
2 command to configure region 2 with the following parameters:





Set the Authoritative Domain field to match the SIP domain of the enterprise. In this
configuration, the domain name is avaya.com. This name appears in the “From” header
of SIP messages originating from this IP region.
Enter a descriptive name in the Name field.
Enable IP-IP Direct Audio (shuffling) to allow audio traffic to be sent directly between
IP endpoints without using media resources in the Avaya Media Gateway. Set both
Intra-region and Inter-region IP-IP Direct Audio to yes. This is the default setting.
Shuffling can be further restricted at the trunk level on the Signaling Group form.
Set the Codec Set field to the IP codec set defined in Section 5.4.
Default values can be used for all other fields.
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change ip-network-region 2
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IP NETWORK REGION
Region: 2
Location: 1
Authoritative Domain: avaya.com
Name: SingTel SIP Trunk
MEDIA PARAMETERS
Intra-region IP-IP Direct Audio: yes
Codec Set: 2
Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048
IP Audio Hairpinning? n
UDP Port Max: 3329
DIFFSERV/TOS PARAMETERS
Call Control PHB Value: 46
Audio PHB Value: 46
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
On Page 4, define the IP codec set to be used for traffic between region 2 and region 1. Enter the
desired IP codec set in the codec set column of the row with destination region (dst rgn) 1.
Default values may be used for all other fields. The example below shows the settings used for
the compliance test. It indicates that codec set 2 will be used for calls between region 2 (the
service provider region) and region 1 (the rest of the enterprise). Creating this table entry for ip
network region 2 will automatically create a complementary table entry on the ip network region
1 form for destination region 2. This complementary table entry can be viewed using the display
ip-network-region 1 command and navigating to Page 4.
change ip-network-region 2
Source Region: 2
Inter Network Region Connection Management
dst codec direct
WAN-BW-limits
Video
Intervening
rgn set
WAN Units
Total Norm Prio Shr Regions
1
2
y
NoLimit
2
2
3
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5.6. Signaling Group
Use the add signaling-group command to create a signaling group between Communication
Manager and Session Manager for use by the service provider trunk. This signaling group is used
for inbound and outbound calls between the service provider and the enterprise. For the
compliance test, signaling group 4 was used for this purpose and was configured using the
parameters highlighted below.












Set the Group Type field to sip.
Set the Transport Method to the recommended default value of tls (Transport Layer
Security). The transport method specified here is used between the Communication
Manager and Session Manager.
Set the IMS Enabled field to n. This specifies the Communication Manager will serve as
an Evolution Server for Session Manager.
Set the Near-end Listen Port and Far-end Listen Port to a valid unused port instead of
the default well-known port value. (For TLS, the well-known port value is 5061 and for
TCP the well-known port value is 5060). At the time of Session Manager installation, a
SIP connection between Communication Manager and Session Manager would have been
established for use by all Communication Manager SIP traffic using the well-known port
value for TLS or TCP. By creating a new signaling group with a separate port value, a
separate SIP connection is created between Communication Manager and Session
Manager for SIP traffic to the service provider. As a result, any signaling group or trunk
group settings (Section 5.7) will only affect the service provider traffic and not other SIP
traffic at the enterprise. The compliance test was conducted with the Near-end Listen
Port and Far-end Listen Port set to 5062.
Set the Peer Detection Enabled field to y. The Peer-Server field will initially be set to
Others and can not be changed via administration. Later, the Peer-Server field will
automatically change to SM once Communication Manager detects its peer as a Session
Manager.
Set the Near-end Node Name to procr. This node name maps to the IP address of
Communication Manager running on the Avaya S8800 Server as defined in Section 5.3.
Set the Far-end Node Name to SM. This node name maps to the IP address of Session
Manager as defined in Section 5.3.
Set the Far-end Network Region to the IP network region defined for the service
provider in Section 5.5.
Set the Far-end Domain to the domain of the enterprise.
Set Direct IP-IP Audio Connections to y. This field will enable media shuffling on the
SIP trunk allowing Communication Manager to redirect media traffic directly between
the SIP trunk and the enterprise endpoint.
Set the DTMF over IP field to rtp-payload. This value enables Communication
Manager to send DTMF transmissions using RFC 2833.
Set the Alternate Route Timer to 6. This defines the number of seconds the that
Communication Manager will wait for a response (other than 100 Trying) to an outbound
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
INVITE before selecting another route. If an alternate route is not defined, then the call is
cancelled after this interval.
Default values may be used for all other fields.
change signaling-group 4
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SIGNALING GROUP
Group Number: 4
Group Type: sip
IMS Enabled? n
Transport Method: tls
Q-SIP? n
IP Video? n
Peer Detection Enabled? y Peer Server: SM
Near-end Node Name: procr
Near-end Listen Port: 5062
SIP Enabled LSP? n
Enforce SIPS URI for SRTP? y
Far-end Node Name: SM
Far-end Listen Port: 5062
Far-end Network Region: 2
Far-end Secondary Node Name:
Far-end Domain: avaya.com
Incoming Dialog Loopbacks: eliminate
DTMF over IP: rtp-payload
Session Establishment Timer(min): 3
Enable Layer 3 Test? y
Bypass If IP Threshold Exceeded?
RFC 3389 Comfort Noise?
Direct IP-IP Audio Connections?
IP Audio Hairpinning?
n
n
y
n
Alternate Route Timer(sec): 6
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5.7. Trunk Group
Use the add trunk-group command to create a trunk group for the signaling group created in
Section 5.6. For the compliance test, trunk group 4 was configured using the parameters
highlighted below.








Set the Group Type field to sip.
Enter a descriptive name for the Group Name.
Enter an available trunk access code (TAC) that is consistent with the existing dial plan
in the TAC field.
Set the Service Type field to public-ntwrk.
Set Member Assignment Method to auto.
Set the Signaling Group to the signaling group shown in the previous step.
Set the Number of Members field to the number of trunk members in the SIP trunk
group. This value determines how many simultaneous SIP calls can be supported by this
trunk.
Default values were used for all other fields.
change trunk-group 4
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TRUNK GROUP
Group Number:
Group Name:
Direction:
Dial Access?
Queue Length:
Service Type:
4
Group Type:
SingTel SIP Trunk
COR:
two-way
Outgoing Display?
n
0
public-ntwrk
Auth Code?
sip
CDR Reports: y
1
TN: 1
TAC: #04
n
Night Service:
n
Member Assignment Method: auto
Signaling Group: 4
Number of Members: 255
Verify that the Preferred Minimum Session Refresh Interval is set to a value acceptable to the
service provider. This value defines the interval that re-INVITEs must be sent to keep the active
session alive. For the compliance test, the value of 600 seconds was used.
change trunk-group 4
Group Type: sip
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TRUNK PARAMETERS
Unicode Name: auto
Redirect On OPTIM Failure: 5000
SCCAN? n
Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 600
Disconnect Supervision - In? y
XOIP Treatment: auto
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Out? y
Delay Call Setup When Accessed Via IGAR? n
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On Page 3, set the Numbering Format field to public. This field specifies the format of the
calling party number (CPN) sent to the far-end. Beginning with Communication Manager 6.0,
public numbers are automatically preceded with a + sign (E.164 format) when passed in the SIP
“From”, “Contact” and “P-Asserted Identity” headers. The addition of the + sign does not
impact interoperability with SingTel.
Set the Replace Restricted Numbers and Replace Unavailable Numbers fields to y. This will
allow the CPN displayed on local endpoints to be replaced with the value set in Section 5.2, if
the inbound call enabled CPN block. For outbound calls, these same settings request that CPN
block be activated on the far-end destination if a local user requests CPN block on a particular
call routed out this trunk. Default values were used for all other fields.
change trunk-group 4
TRUNK FEATURES
ACA Assignment? n
Page
3 of
21
Measured: none
Maintenance Tests? y
Numbering Format: public
UUI Treatment: service-provider
Replace Restricted Numbers? y
Replace Unavailable Numbers? y
Modify Tandem Calling Number: no
Show ANSWERED BY on Display? n
DSN Term? n
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On Page 4, set the Network Call Redirection field to n. Set the Send Diversion Header field
to y and the Support Request History field to n. The Send Diversion Header field provides
additional information to the network if the call has been re-directed. These settings are needed
to support call forwarding of inbound calls back to the PSTN and some Extension to Cellular
(EC500) call scenarios.
Set the Telephone Event Payload Type to 101, the value preferred by SingTel.
change trunk-group 4
Page
4 of
21
PROTOCOL VARIATIONS
Mark Users as Phone?
Prepend '+' to Calling Number?
Send Transferring Party Information?
Network Call Redirection?
Send Diversion Header?
Support Request History?
Telephone Event Payload Type:
Convert 180 to 183 for Early Media?
Always Use re-INVITE for Display Updates?
Identity for Calling Party Display:
Enable Q-SIP?
n
n
n
n
y
n
101
n
n
P-Asserted-Identity
n
5.8. Calling Party Information
The calling party number is sent in the SIP “From”, “Contact” and “PAI” headers. Since public
numbering was selected to define the format of this number (Section 5.7), use the change
public-numbering command to create an entry for each extension which has a DID assigned.
The DID number will be one assigned by the SIP service provider. It is used to authenticate the
caller.
In the sample configuration, three DID numbers were assigned for testing. These three numbers
were assigned to the three extensions 40001, 40010 and 40022. Thus, these same 10-digit
numbers were used in the outbound calling party information on the service provider trunk when
calls were originated from these three extensions.
change public-unknown-numbering 0
Page
1 of
2
NUMBERING - PUBLIC/UNKNOWN FORMAT
Total
Ext Ext
Trk
CPN
CPN
Len Code
Grp(s)
Prefix
Len
Total Administered: 9
5 1
5
Maximum Entries: 9999
5 4
5
5 40001
4
6568596345
10
Note: If an entry applies to
5 40010
4
6568596346
10
a SIP connection to Avaya
5 40022
4
6568596343
10
Aura(tm) Session Manager,
the resulting number must
be a complete E.164 number.
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5.9. Outbound Call Routing
In these Application Notes, the Automatic Route Selection (ARS) feature is used to route
outbound calls via the SIP trunk to the service provider. In the sample configuration, the single
digit 9 is used as the ARS access code. Enterprise callers will dial 9 to reach an “outside line”.
This common configuration is illustrated below with little elaboration. Use the change dialplan
analysis command to define a dialed string beginning with 9 of length 1 as a feature access code
(fac).
change dialplan analysis
Page
DIAL PLAN ANALYSIS TABLE
Location: all
Dialed
String
0
4
9
*
#
Total Call
Length Type
1
attd
5
ext
1
fac
3
fac
3
dac
Dialed
String
Total Call
Length Type
1 of
12
Percent Full: 2
Dialed
String
Total Call
Length Type
Use the change feature-access-codes command to configure 9 as the Auto Route Selection
(ARS) – Access Code 1.
change feature-access-codes
Page
1 of
11
FEATURE ACCESS CODE (FAC)
Abbreviated Dialing List1 Access Code: *10
Abbreviated Dialing List2 Access Code: *12
Abbreviated Dialing List3 Access Code: *13
Abbreviated Dial - Prgm Group List Access Code: *14
Announcement Access Code: *19
Answer Back Access Code:
Auto Alternate Routing (AAR) Access Code:
Auto Route Selection (ARS) - Access Code 1:
Automatic Callback Activation:
Call Forwarding Activation Busy/DA: *30
All:
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*00
9
*33
*31
Access Code 2:
Deactivation: #33
Deactivation: #30
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Use the change ars analysis command to configure the routing of dialed digits following the
first digit 9. The example below shows a subset of the dialed strings tested as part of the
compliance test. See Section 2.1 for the complete list of call types tested. All dialed strings are
mapped to route pattern 4 which contains the SIP trunk to the service provider (as defined next).
change ars analysis 0
Page
ARS DIGIT ANALYSIS TABLE
Location: all
Dialed
String
Total
Min Max
11
23
8
8
8
8
8
8
8
8
001
3
6
8
9
Route
Pattern
4
4
4
4
4
Call
Type
intl
pubu
pubu
pubu
pubu
1 of
2
Percent Full: 0
Node
Num
ANI
Reqd
n
n
n
n
n
The route pattern defines which trunk group will be used for the call and performs any necessary
digit manipulation. Use the change route-pattern command to configure the parameters for the
service provider trunk route pattern in the following manner. The example below shows the
values used for route pattern 4 during the compliance test.



Pattern Name: Enter a descriptive name.
Grp No: Enter the outbound trunk group for the SIP service provider. For the compliance
test, trunk group 4 was used.
FRL: Set the Facility Restriction Level (FRL) field to a level that allows access to this
trunk for all users that require it. The value of 0 is the least restrictive level.
change route-pattern 4
Page
Pattern Number: 4
Pattern Name: SingTelSIPTrunk
SCCAN? n
Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted
No
Mrk Lmt List Del Digits
Dgts
1: 4
0
2:
3:
4:
5:
6:
1:
2:
3:
4:
5:
6:
BCC VALUE TSC CA-TSC
0 1 2 M 4 W
Request
ITC BCIE Service/Feature PARM
y
y
y
y
y
y
rest
rest
rest
rest
rest
rest
y
y
y
y
y
y
y
y
y
y
y
y
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y
y
y
y
y
y
y
y
y
y
y
y
n
n
n
n
n
n
n
n
n
n
n
n
1 of
3
DCS/
QSIG
Intw
n
n
n
n
n
n
IXC
user
user
user
user
user
user
No. Numbering LAR
Dgts Format
Subaddress
none
none
none
none
none
none
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5.10. Inbound Call Routing
This step configures the settings necessary to map incoming DID calls to the proper
Communication Manager extension(s). The incoming digits sent in the INVITE message from
SingTel can be manipulated as necessary to route calls to the desired extension. In general, the
DID numbers should correlate with the internal extensions, so that only some of the leading
digits need to be deleted. However, for this testing, all the DID digits were deleted and replaced
by the internal extension as illustrated below.
change inc-call-handling-trmt trunk-group 4
INCOMING CALL HANDLING TREATMENT
Service/
Number
Number
Del Insert
Feature
Len
Digits
public-ntwrk
11 +6568596343
11 40001
public-ntwrk
11 +6568596345
11 40022
public-ntwrk
11 +6568596346
11 40010
Page
1 of
1
Per Call Night
CPN/BN
Serv
6. Configure Avaya Aura® Session Manager
This section provides the procedures for configuring Session Manager. The procedures include
adding the following items:





SIP domain.
Logical/physical Location that can be occupied by SIP Entities.
Adaptation module to perform dial plan manipulation.
SIP Entities corresponding to Communication Manager, the SBC and Session Manager.
Entity Links, which define the SIP trunk parameters used by Session Manager when routing
calls to/from SIP Entities.
 Routing Policies, which control call routing between the SIP Entities.
 Dial Patterns, which govern to which SIP Entity a call is routed.
 Session Manager, to be managed by System Manager.
It may not be necessary to create all the items above when creating a connection to the service
provider since some of these items would have already been defined as part of the initial Session
Manager installation. This includes items such as SIP domains, locations, SIP entities, and
Session Manager itself. However, each item should be reviewed to verify the configuration.
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6.1. Avaya Aura® System Manager Login and Navigation
Session 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 and click on Login (not shown).
The Home Screen as shown below is then displayed. Select Routing under Elements Section.
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The navigation tree displayed in the left pane below will be referenced in subsequent sections to
navigate to items requiring configuration.
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6.2. Specify SIP Domain
Create a SIP domain for each domain for which Session Manager will need to be aware in order
to route calls. For the compliance test, this includes the enterprise domain (avaya.com).
Navigate to Routing  Domains in the left-hand navigation pane (Section 6.1) and click the
New button in the right pane (not shown). In the new right pane that appears (shown below), fill
in the following:
 Name:
 Type:
 Notes:
Enter the domain name.
Select sip from the pull-down menu.
Add a brief description (optional).
Click Commit. The screen below shows the entry for the enterprise domain.
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6.3. Configure Location
Locations can be used to identify logical and/or physical locations where SIP Entities reside for
purposes of bandwidth management and call admission control. For the compliance test, one
location was defined based on the enterprise IP subnet shown in Figure 1. To add a location,
navigate to Routing  Locations in the left-hand navigation pane and click the New button in
the right pane (not shown). In the new right pane that appears (shown below), fill in the
following:
In the General section, enter the following values. Use default values for all remaining fields.
 Name: Enter a descriptive name for the location.
 Notes: Add a brief description (optional).
In the Location Pattern section, click Add and enter the following values. Use default values
for all remaining fields.
 IP Address Pattern:
An IP address pattern used to identify the location.
 Notes:
Add a brief description (optional).
The screen below shows the addition of the location named TestLocation, which includes all
equipment on the 10.1.40.x IP subnet including Communication Manager, and the SBC. Click
Commit to save.
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6.4. Add Adaptation Module
Session Manager can be configured with adaptation modules that can modify SIP messages
before or after routing decisions have been made. A generic adaptation module
DigitConversionAdapter supports digit conversion of telephone numbers in specific headers of
SIP messages. Other adaptation modules are built on this generic, and can modify other headers
to permit interoperability with third party SIP products.
For SingTel interoperability, one adaptation is needed. The adaptation is applied to the Acme
Packet SBC SIP entity and converts the domain part of the outbound Request URI header from
Session Manager containing the enterprise domain to the SingTel SIP proxy IP address.
To create the adaptation, navigate to Routing  Adaptations in the left-hand navigation pane
and click on the New button in the right pane (not shown). In the new right pane that appears
(shown below), fill in the following:
In the General section, enter the following values. Use default values for all remaining fields.
 Adaptation name: Enter a descriptive name for the adaptation.
 Module name:
Enter DigitConversionAdapter.
 Module parameter: Enter fromto=true odstd=<ipaddr>, where <ipaddr> is the IP
address of the SBC located on SingTel network. This value is
provided by SingTel. This parameter replaces the domain in the
Request URI header with the given value for outbound only.
 Notes:
Add a brief description (optional).
Click Commit to save.
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6.5.
Add SIP Entities
A SIP Entity must be added for Session Manager and for each SIP telephony system connected
to Session Manager which includes Communication Manager and the SBC. Navigate to Routing
 SIP Entities in the left-hand navigation pane and click on the New button in the right pane
(not shown). In the new right pane that appears (shown below), fill in the following:
In the General section, enter the following values. Use default values for all remaining fields.
 Name:
Enter a descriptive name.
 FQDN or IP Address: Enter the FQDN or IP address of the SIP Entity that is used for SIP
signaling.
 Type:
Enter Session Manager for Session Manager, CM for
Communication Manager and SIP Trunk for the SBC.
 Adaptation:
This field is only present if Type is not set to Session Manager.
If applicable, select the appropriate Adaptation name created in
Section 6.4 that will be applied to this entity.
 Location:
Select the location that applies to the SIP entity being created. For
the compliance test, all SIP Entites are located in TestLocation.
Time Zone:
Select the time zone for the location above.
The following screen shows the addition of Session Manager. The IP address of the virtual SM100 Security Module is entered for FQDN or IP Address.
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To define the ports used by Session Manager, scroll down to the Port section of the SIP Entity
Details screen. This section is only present for Session Manager SIP entities.
In the Port section, click Add and enter the following values. Use default values for all
remaining fields:
 Port:
Port number on which the Session Manager can listen for SIP
requests.
 Protocol:
Transport protocol to be used with this port.
 Default Domain:
The default domain associated with this port. For the compliance
test, this was the enterprise SIP domain.
Defaults can be used for the remaining fields. Click Commit to save.
For the compliance test, four Port entries were added as shown below.
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The following screen shows the addition of Communication Manager. In order for Session
Manager to send SIP service provider traffic on a separate entity link to Communication
Manager, this requires the creation of a separate SIP entity for Communication Manager than the
one created at Session Manager installation for use with all other SIP traffic. The FQDN or IP
Address field is set to the IP address of Communication Manager running on the Avaya S8800
Server. The Location field is set to TestLocation which is the location defined for the subnet
where Communication Manager resides.
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The following screen shows the addition of the Acme Packet SBC. The FQDN or IP Address
field is set to the IP address of its private network interface (see Figure 1). For the Adaptation
field, select the adaptation module previously defined for this SIP entity in Section 6.4. The
Location field is set to TestLocation which is the location defined for the subnet where the SBC
resides.
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6.6. Add Entity Links
A SIP trunk between Session Manager and a telephony system is described by an Entity Link.
Two Entity Links were created: one to Communication Manager and one to the SBC. To add an
Entity Link, navigate to Routing  Entity Links in the left-hand navigation pane and click on
the New button in the right pane (not shown). In the new right pane that appears (shown below),
fill in the following:




Name:
SIP Entity 1:
Protocol:
Port:
 SIP Entity 2:
 Port:
 Trusted:
Enter a descriptive name.
Select the Session Manager.
Select the transport protocol used for this link.
Port number on which Session Manager will receive SIP requests from
the far-end. For the Communication Manager Entity Link, this must
match the Far-end Listen Port defined on the Communication Manager
signaling group in Section 5.6.
Select the name of the other system. For the Communication Manager
Entity Link, select the Communication Manager SIP Entity defined in
Section 6.5.
Port number on which the other system receives SIP requests from the
Session Manager. For the Communication Manager Entity Link, this must
match the Near-end Listen Port defined on the Communication Manager
signaling group in Section 5.6.
Check this box. Note: If this box is not checked, calls from the associated
SIP Entity specified in Section 6.5 will be denied.
Click Commit to save. The following screen illustrates the Entity Link to Communication
Manager. The protocol and ports defined here must match the values used on the
Communication Manager signaling group form in Section 5.6.
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The following screen illustrates the Entity Link between Session Manager and the Acme Packet
SBC.
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6.7. Add Routing Policies
Routing policies describe the conditions under which calls will be routed to the SIP Entities
specified in Section 6.5. Two routing policies must be added: one for Communication Manager
and one for the SBC. To add a routing policy, navigate to Routing Routing Policies in the
left-hand navigation pane and click on the New button in the right pane (not shown). In the new
right pane that appears (shown below), fill in the following:
In the General section, enter the following values. Use default values for all remaining fields.
 Name:
Enter a descriptive name.
 Notes:
Add a brief description (optional).
In the SIP Entity as Destination section, click Select. The SIP Entity List page opens (not
shown). Select the appropriate SIP entity to which this routing policy applies and click Select.
The selected SIP Entity displays on the Routing Policy Details page as shown below. Use default
values for remaining fields. Click Commit to save. The following screens show the Routing
Policies for Communication Manager and the SBC.
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6.8. Add Dial Patterns
Dial Patterns are needed to route calls through Session Manager. For the compliance test, dial
patterns were needed to route calls from Communication Manager to SingTel and vice versa.
Dial Patterns define which route policy will be selected for a particular call based on the dialed
digits, destination domain and originating location. To add a dial pattern, navigate to Routing
Dial Patterns in the left-hand navigation pane and click on the New button in the right pane
(not shown). In the new right pane that appears (shown below), fill in the following:
In the General section, enter the following values. Use default values for all remaining fields.
 Pattern:
Enter a dial string that will be matched against the Request-URI of the call.
 Min:
Enter a minimum length used in the match criteria.
 Max:
Enter a maximum length used in the match criteria.
 SIP Domain: Enter the destination domain used in the match criteria.
 Notes:
Add a brief description (optional).
In the Originating Locations and Routing Policies section, click Add. From the Originating
Locations and Routing Policy List that appears (not shown), select the appropriate originating
location for use in the match criteria. Lastly, select the routing policy from the list that will be
used to route all calls that match the specified criteria. Click Select.
Default values can be used for the remaining fields. Click Commit to save.
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Two examples of the dial patterns used for the compliance test are shown below. The first
example shows that 8-digit numbers that begin with a 6 and have a destination domain of
avaya.com from Any Locations uses route policy To-AcmeSBC-PSTN.
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The second example shows that 11-digit numbers (including the + sign) that start with
+656859634 to domain avaya.com and originating from Any Locations uses route policy ToCM. These are the DID numbers assigned to the enterprise from SingTel.
The complete list of dial patterns defined for the compliance test is shown below.
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6.9. Add/View Session Manager
The creation of a Session Manager element provides the linkage between System Manager and
Session Manager. This was most likely done as part of the initial Session Manager installation.
To add a Session Manager, navigate to Elements  Session Manager Session Manager
Administration from the Home Screen and click on the New button in the right pane (not
shown). If the Session Manager already exists, select the appropriate Session Manager and click
View (not shown) to view the configuration. Enter/verify the data as described below and shown
in the following screen:
In the General section, enter the following values:
 SIP Entity Name:
Select the SIP Entity created for Session
Manager.
 Description:
Add a brief description (optional).
 Management Access Point Host Name/IP: Enter the IP address of the Session Manager
management interface.
The screen below shows the Session Manager values used for the compliance test.
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In the Security Module section, enter the following values:
 SIP Entity IP Address:
Should be filled in automatically based on the SIP Entity
Name. Otherwise, enter IP address of Session Manager
signaling interface.
 Network Mask:
Enter the network mask corresponding to the IP address of
Session Manager.
 Default Gateway:
Enter the IP address of the default gateway for Session
Manager.
Use default values for the remaining fields. Click Save (not shown) to add this Session
Manager. The screen below shows the remaining Session Manager values used for the
compliance test.
7. Configure Acme Packet 4250 Net-Net Session Border
Controller
The Acme Packet 4250 Net-Net SBC was installed and provisioned by Acme Packet engineers
for this testing. As such, the step-by-step provisioning of the SBC is not discussed in these
Application Notes. The SBC configuration file is shown in Appendix A for reference.
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8. SingTel Meg@POP SIP Trunking Service Configuration
In order to use SingTel SIP Trunking Service on the Meg@POP IP VPN Network, a customer
must order the service from SingTel. For further information on SingTel Meg@POP as well as
its network and access services, contact a SingTel Account Manager or call 1800-763-1111
(local toll-free).
SingTel will provide the IP address of the SingTel SIP proxy/SBC, IP addresses of media
sources and Direct Inward Dialed (DID) numbers assigned to the enterprise. This information is
used to configure Communication Manager, Session Manager, and Acme Packet SBC discussed
in the previous sections.
The configuration between SingTel Meg@POP SIP Trunking Service and the enterprise is a
static configuration. There is no registration of the SIP trunk or enterprise users to the SingTel
network.
9. Verification Steps
This section provides verification steps that may be performed in the field to verify that the
solution is configured properly. This section also provides a list of useful troubleshooting
commands that can be used to troubleshoot the solution.
Verification Steps:
1. Verify that endpoints at the enterprise site can place calls to the PSTN and that the call
remains active for more than 35 seconds. This time period is included to verify that proper
routing of the SIP messaging has satisfied SIP protocol timers.
2. Verify that endpoints at the enterprise site can receive calls from the PSTN and that the call
can remain active for more than 35 seconds.
3. Verify that the user on the PSTN can end an active call by hanging up.
4. Verify that an endpoint at the enterprise site can end an active call by hanging up.
Troubleshooting:
1. Communication Manager:
 list trace station <extension number> - Traces calls to and from a specific station.
 list trace tac <trunk access code> - Traces calls over a specific trunk group.
 status station <extension number> - Displays signaling and media information for an
active call on a specific station.
 status trunk <trunk-group number> - Displays trunk group information.
 status trunk <trunk-group number/member-number> - Displays signaling and media
information for an active trunk member.
2. Session Manager:
 Call Routing Test - The Call Routing Test verifies the routing for a particular source and
destination. To run the routing test, navigate to Elements > Session Manager > System
Tools > Call Routing Test. Enter the requested data to run the test.
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10. Conclusion
These Application Notes describe the configuration necessary to connect Avaya Aura®
Communication Manager 6.0.1, Avaya Aura® Session Manager 6.1 and Acme Packet 4250 NetNet Session Border Controller to SingTel Meg@POP SIP Trunking Service. SingTel Meg@POP
SIP Trunking Service passed compliance testing. Please refer to Section 2.2 for any exceptions
or workarounds.
11. Additional References
This section references the documentation relevant to these Application Notes. Additional
Avaya product documentation is available at http://support.avaya.com.
[1] Administering Avaya Aura® Communication Manager, Release 6.0, June 2010, Document
Number 03-300509, Issue 6.0.
[2] Avaya Aura® Communication Manager Feature Description and Implementation, Release
6.0, June 2010, Document Number 555-245-205, Issue 8.0.
[3] Administering Avaya Aura® System Manager, Release 6.1, November 2010.
[4] Administering Avaya Aura® Session Manager, Release 6.1, November 2010, Document
Number 03-603324, Issue 1.
[5] Avaya 1600 Series IP Deskphones Administrator Guide Release 1.3.x, May 2010, Document
Number 16-601443.
[6] Avaya one-X® Deskphone Edition for 9600 Series IP Telephones Administrator Guide,
November 2009, Document Number 16-603838, Issue 1.
[7] Avaya one-X™ Deskphone SIP Administrator Guide, December 2010, Document Number
16-300698.
[8] Using Avaya one-X® Communicator Release 6.1, April 2011.
[9] RFC 3261 SIP: Session Initiation Protocol, http://www.ietf.org/
[10] RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals,
http://www.ietf.org/
[11] RFC 4244, An Extension to the Session Initiation Protocol (SIP) for Request History
Information, http://www.ietf.org/
[12] RFC 4856, Media Type Registration of Payload Formats in the RTP Profile for Audio and
Video Conferences, http://www.ietf.org/
JC; Reviewed:
SPOC 11/8/2011
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12. Appendix A: Acme Packet 4250 Net-Net Session Border
Controller Configuration File
local-policy
from-address
*
to-address
*
source-realm
access
description
activate-time
deactivate-time
state
policy-priority
last-modified-by
last-modified-date
policy-attribute
next-hop
realm
action
terminate-recursion
carrier
start-time
end-time
days-of-week
cost
app-protocol
state
methods
media-profiles
local-policy
from-address
N/A
N/A
enabled
none
admin@console
2008-09-24 17:07:15
10.1.40.24
core
none
disabled
0000
2400
U-S
0
SIP
enabled
*
to-address
*
source-realm
core
description
activate-time
deactivate-time
state
policy-priority
last-modified-by
last-modified-date
policy-attribute
next-hop
realm
action
terminate-recursion
carrier
start-time
end-time
days-of-week
JC; Reviewed:
SPOC 11/8/2011
N/A
N/A
enabled
none
admin@console
2008-10-04 15:46:15
x.x.x.x
(SingTel SBC)
none
disabled
0000
2400
U-S
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cost
0
app-protocol
SIP
state
enabled
methods
media-profiles
media-manager
state
enabled
latching
enabled
flow-time-limit
86400
initial-guard-timer
300
subsq-guard-timer
300
tcp-flow-time-limit
86400
tcp-initial-guard-timer
300
tcp-subsq-guard-timer
300
tcp-number-of-ports-per-flow
2
hnt-rtcp
disabled
algd-log-level
NOTICE
mbcd-log-level
NOTICE
red-flow-port
1985
red-mgcp-port
1986
red-max-trans
10000
red-sync-start-time
5000
red-sync-comp-time
1000
media-policing
enabled
max-signaling-bandwidth
10000000
max-untrusted-signaling
100
min-untrusted-signaling
30
app-signaling-bandwidth
0
tolerance-window
30
rtcp-rate-limit
0
min-media-allocation
32000
min-trusted-allocation
1000
deny-allocation
1000
anonymous-sdp
disabled
arp-msg-bandwidth
32000
fragment-msg-bandwidth
0
rfc2833-timestamp
disabled
default-2833-duration
100
rfc2833-end-pkts-only-for-non-sig enabled
translate-non-rfc2833-event
disabled
dnsalg-server-failover
disabled
last-modified-by
admin@console
last-modified-date
2007-09-22 10:08:00
network-interface
name
M10
sub-port-id
0
description
hostname
ip-address
10.1.40.30
pri-utility-addr
sec-utility-addr
netmask
255.255.255.0
gateway
10.1.40.24
sec-gateway
gw-heartbeat
state
disabled
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heartbeat
retry-count
retry-timeout
health-score
dns-ip-primary
dns-ip-backup1
dns-ip-backup2
dns-domain
dns-timeout
hip-ip-list
ftp-address
icmp-address
snmp-address
telnet-address
last-modified-by
last-modified-date
network-interface
name
sub-port-id
description
hostname
ip-address
pri-utility-addr
sec-utility-addr
netmask
gateway
sec-gateway
gw-heartbeat
state
heartbeat
retry-count
retry-timeout
health-score
dns-ip-primary
dns-ip-backup1
dns-ip-backup2
dns-domain
dns-timeout
hip-ip-list
ftp-address
icmp-address
snmp-address
telnet-address
last-modified-by
last-modified-date
phy-interface
name
operation-type
port
slot
virtual-mac
admin-state
auto-negotiation
duplex-mode
speed
last-modified-by
JC; Reviewed:
SPOC 11/8/2011
0
0
1
0
11
10.1.40.30
10.1.40.30
admin@console
2008-09-24 16:45:01
M00
0
192.168.3.2
255.255.255.0
192.168.3.11
disabled
0
0
1
0
11
192.168.3.2
192.168.3.2
192.168.3.2
192.168.3.2
admin@10.1.1.188
2008-09-24 18:05:45
M00
Media
0
0
enabled
enabled
FULL
100
admin@console
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last-modified-date
phy-interface
name
operation-type
port
slot
virtual-mac
admin-state
auto-negotiation
duplex-mode
speed
last-modified-by
last-modified-date
realm-config
identifier
description
addr-prefix
network-interfaces
mm-in-realm
mm-in-network
mm-same-ip
mm-in-system
bw-cac-non-mm
msm-release
qos-enable
generate-UDP-checksum
max-bandwidth
fallback-bandwidth
max-priority-bandwidth
max-latency
max-jitter
max-packet-loss
observ-window-size
parent-realm
dns-realm
media-policy
in-translationid
out-translationid
in-manipulationid
out-manipulationid
manipulation-string
class-profile
average-rate-limit
access-control-trust-level
invalid-signal-threshold
maximum-signal-threshold
untrusted-signal-threshold
nat-trust-threshold
deny-period
ext-policy-svr
symmetric-latching
pai-strip
trunk-context
early-media-allow
enforcement-profile
JC; Reviewed:
SPOC 11/8/2011
2007-09-22 10:04:09
M10
Media
0
1
enabled
enabled
FULL
100
admin@console
2007-09-22 10:04:31
core
0.0.0.0
M10:0
disabled
enabled
enabled
enabled
disabled
disabled
disabled
disabled
0
0
0
0
0
0
0
0
none
0
0
0
0
30
disabled
disabled
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additional-prefixes
restricted-latching
restriction-mask
accounting-enable
user-cac-mode
user-cac-bandwidth
user-cac-sessions
icmp-detect-multiplier
icmp-advertisement-interval
icmp-target-ip
monthly-minutes
net-management-control
delay-media-update
refer-call-transfer
codec-policy
codec-manip-in-realm
constraint-name
call-recording-server-id
stun-enable
stun-server-ip
stun-server-port
stun-changed-ip
stun-changed-port
match-media-profiles
qos-constraint
last-modified-by
last-modified-date
realm-config
identifier
description
addr-prefix
network-interfaces
mm-in-realm
mm-in-network
mm-same-ip
mm-in-system
bw-cac-non-mm
msm-release
qos-enable
generate-UDP-checksum
max-bandwidth
fallback-bandwidth
max-priority-bandwidth
max-latency
max-jitter
max-packet-loss
observ-window-size
parent-realm
dns-realm
media-policy
in-translationid
out-translationid
in-manipulationid
out-manipulationid
manipulation-string
JC; Reviewed:
SPOC 11/8/2011
none
32
enabled
none
0
0
0
0
0
disabled
disabled
disabled
disabled
disabled
0.0.0.0
3478
0.0.0.0
3479
admin@console
2007-09-22 10:11:43
access
0.0.0.0
M00:0
disabled
enabled
enabled
enabled
disabled
disabled
disabled
disabled
0
0
0
0
0
0
0
NAT_IP
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class-profile
average-rate-limit
access-control-trust-level
invalid-signal-threshold
maximum-signal-threshold
untrusted-signal-threshold
nat-trust-threshold
deny-period
ext-policy-svr
symmetric-latching
pai-strip
trunk-context
early-media-allow
enforcement-profile
additional-prefixes
restricted-latching
restriction-mask
accounting-enable
user-cac-mode
user-cac-bandwidth
user-cac-sessions
icmp-detect-multiplier
icmp-advertisement-interval
icmp-target-ip
monthly-minutes
net-management-control
delay-media-update
refer-call-transfer
codec-policy
codec-manip-in-realm
constraint-name
call-recording-server-id
stun-enable
stun-server-ip
stun-server-port
stun-changed-ip
stun-changed-port
match-media-profiles
qos-constraint
last-modified-by
last-modified-date
session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
JC; Reviewed:
SPOC 11/8/2011
0
none
0
0
0
0
30
disabled
disabled
none
32
enabled
none
0
0
0
0
0
disabled
disabled
disabled
disabled
disabled
0.0.0.0
3478
0.0.0.0
3479
admin@console
2008-09-22 16:58:57
x.x.x.x
x.x.x.x
5060
enabled
SIP
(SingTel SBC)
(SingTel SBC)
UDP
access
enabled
disabled
0
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max-inbound-sessions
max-outbound-sessions
max-burst-rate
max-inbound-burst-rate
max-outbound-burst-rate
max-sustain-rate
max-inbound-sustain-rate
max-outbound-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
ping-send-mode
ping-in-service-response-codes
out-service-response-codes
media-profiles
in-translationid
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
manipulation-string
p-asserted-id
trunk-group
max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
codec-policy
enforcement-profile
refer-call-transfer
reuse-connections
tcp-keepalive
tcp-reconn-interval
max-register-burst-rate
register-burst-window
last-modified-by
last-modified-date
JC; Reviewed:
SPOC 11/8/2011
0
0
0
0
0
0
0
0
5
0
0
0
0
0
0
None
enabled
enabled
0
keep-alive
disabled
disabled
0
disabled
none
0
disabled
NONE
none
0
0
0
admin@console
2008-10-04 15:47:19
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session-agent
hostname
ip-address
port
state
app-protocol
app-type
transport-method
realm-id
egress-realm-id
description
carriers
allow-next-hop-lp
constraints
max-sessions
max-inbound-sessions
max-outbound-sessions
max-burst-rate
max-inbound-burst-rate
max-outbound-burst-rate
max-sustain-rate
max-inbound-sustain-rate
max-outbound-sustain-rate
min-seizures
min-asr
time-to-resume
ttr-no-response
in-service-period
burst-rate-window
sustain-rate-window
req-uri-carrier-mode
proxy-mode
redirect-action
loose-routing
send-media-session
response-map
ping-method
ping-interval
ping-send-mode
ping-in-service-response-codes
out-service-response-codes
media-profiles
in-translationid
out-translationid
trust-me
request-uri-headers
stop-recurse
local-response-map
ping-to-user-part
ping-from-user-part
li-trust-me
in-manipulationid
out-manipulationid
manipulation-string
p-asserted-id
trunk-group
JC; Reviewed:
SPOC 11/8/2011
10.1.40.24
10.1.40.24
5060
enabled
SIP
StaticTCP
core
enabled
disabled
0
0
0
0
0
0
0
0
0
5
0
0
0
0
0
0
None
enabled
enabled
0
keep-alive
disabled
disabled
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max-register-sustain-rate
early-media-allow
invalidate-registrations
rfc2833-mode
rfc2833-payload
codec-policy
enforcement-profile
refer-call-transfer
reuse-connections
tcp-keepalive
tcp-reconn-interval
max-register-burst-rate
register-burst-window
last-modified-by
last-modified-date
sip-config
state
operation-mode
dialog-transparency
home-realm-id
egress-realm-id
nat-mode
registrar-domain
registrar-host
registrar-port
register-service-route
init-timer
max-timer
trans-expire
invite-expire
inactive-dynamic-conn
enforcement-profile
pac-method
pac-interval
pac-strategy
pac-load-weight
pac-session-weight
pac-route-weight
pac-callid-lifetime
pac-user-lifetime
red-sip-port
red-max-trans
red-sync-start-time
red-sync-comp-time
add-reason-header
sip-message-len
enum-sag-match
extra-method-stats
registration-cache-limit
register-use-to-for-lp
options
add-ucid-header
proxy-sub-events
last-modified-by
last-modified-date
sip-feature
JC; Reviewed:
SPOC 11/8/2011
0
disabled
none
0
disabled
NONE
none
0
0
0
admin@10.1.1.12
2008-09-27 14:51:16
enabled
dialog
enabled
core
None
*
*
5060
always
500
4000
32
180
32
10
PropDist
1
1
1
600
3600
1988
10000
5000
1000
disabled
4096
disabled
disabled
0
disabled
max-udp-length=0
disabled
admin@10.1.1.12
2008-09-27 18:05:58
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name
realm
support-mode-inbound
require-mode-inbound
proxy-require-mode-inbound
support-mode-outbound
require-mode-outbound
proxy-require-mode-outbound
last-modified-by
last-modified-date
sip-interface
state
realm-id
description
sip-port
address
port
transport-protocol
tls-profile
allow-anonymous
ims-aka-profile
carriers
trans-expire
invite-expire
max-redirect-contacts
proxy-mode
redirect-action
contact-mode
nat-traversal
nat-interval
tcp-nat-interval
registration-caching
min-reg-expire
registration-interval
route-to-registrar
secured-network
teluri-scheme
uri-fqdn-domain
trust-mode
max-nat-interval
nat-int-increment
nat-test-increment
sip-dynamic-hnt
stop-recurse
port-map-start
port-map-end
in-manipulationid
out-manipulationid
manipulation-string
sip-ims-feature
operator-identifier
anonymous-priority
max-incoming-conns
per-src-ip-max-incoming-conns
inactive-conn-timeout
untrusted-conn-timeout
JC; Reviewed:
SPOC 11/8/2011
avayaoption
option-1
Pass
Reject
Pass
Pass
Reject
Pass
admin@10.1.1.12
2008-09-27 17:21:01
enabled
core
10.1.40.30
5060
TCP
agents-only
0
0
0
none
none
30
90
enabled
300
3600
disabled
disabled
disabled
all
3600
10
30
disabled
401,407
0
0
disabled
none
0
0
0
0
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network-id
ext-policy-server
default-location-string
charging-vector-mode
charging-function-address-mode
ccf-address
ecf-address
term-tgrp-mode
implicit-service-route
rfc2833-payload
rfc2833-mode
constraint-name
response-map
local-response-map
ims-aka-feature
enforcement-profile
refer-call-transfer
route-unauthorized-calls
tcp-keepalive
add-sdp-invite
add-sdp-profiles
last-modified-by
last-modified-date
sip-interface
state
realm-id
description
sip-port
address
port
transport-protocol
tls-profile
allow-anonymous
ims-aka-profile
carriers
trans-expire
invite-expire
max-redirect-contacts
proxy-mode
redirect-action
contact-mode
nat-traversal
nat-interval
tcp-nat-interval
registration-caching
min-reg-expire
registration-interval
route-to-registrar
secured-network
teluri-scheme
uri-fqdn-domain
trust-mode
max-nat-interval
nat-int-increment
nat-test-increment
sip-dynamic-hnt
JC; Reviewed:
SPOC 11/8/2011
pass
pass
none
disabled
101
transparent
disabled
disabled
none
disabled
admin@10.1.1.12
2008-09-27 18:11:51
enabled
access
192.168.3.2
5060
UDP
agents-only
0
0
0
none
none
30
90
enabled
300
3600
disabled
disabled
disabled
all
3600
10
30
disabled
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stop-recurse
port-map-start
port-map-end
in-manipulationid
out-manipulationid
manipulation-string
sip-ims-feature
operator-identifier
anonymous-priority
max-incoming-conns
per-src-ip-max-incoming-conns
inactive-conn-timeout
untrusted-conn-timeout
network-id
ext-policy-server
default-location-string
charging-vector-mode
charging-function-address-mode
ccf-address
ecf-address
term-tgrp-mode
implicit-service-route
rfc2833-payload
rfc2833-mode
constraint-name
response-map
local-response-map
ims-aka-feature
enforcement-profile
refer-call-transfer
route-unauthorized-calls
tcp-keepalive
add-sdp-invite
add-sdp-profiles
last-modified-by
last-modified-date
sip-manipulation
name
description
header-rule
name
header-name
action
comparison-type
match-value
msg-type
new-value
methods
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
JC; Reviewed:
SPOC 11/8/2011
401,407
0
0
disabled
none
0
0
0
0
pass
pass
none
disabled
101
transparent
disabled
disabled
none
disabled
admin@console
2008-09-24 16:49:46
NAT_IP
From
From
manipulate
case-sensitive
request
From
From
uri-host
replace
ip
case-sensitive
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new-value
header-rule
name
header-name
action
comparison-type
match-value
msg-type
new-value
methods
element-rule
name
parameter-name
type
action
match-val-type
comparison-type
match-value
new-value
last-modified-by
last-modified-date
steering-pool
ip-address
start-port
end-port
realm-id
network-interface
last-modified-by
last-modified-date
steering-pool
ip-address
start-port
end-port
realm-id
network-interface
last-modified-by
last-modified-date
system-config
hostname
description
location
mib-system-contact
mib-system-name
mib-system-location
snmp-enabled
enable-snmp-auth-traps
enable-snmp-syslog-notify
enable-snmp-monitor-traps
enable-env-monitor-traps
snmp-syslog-his-table-length
snmp-syslog-level
system-log-level
process-log-level
process-log-ip-address
process-log-port
collect
JC; Reviewed:
SPOC 11/8/2011
$LOCAL_IP
To
To
manipulate
case-sensitive
request
To
To
uri-host
replace
ip
case-sensitive
$REMOTE_IP
admin@console
2008-09-22 16:56:32
192.168.3.2
20000
20099
access
M00:0
admin@console
2008-09-24 16:54:48
10.1.40.30
20000
20099
core
M10:0
admin@console
2008-09-24 16:55:05
sd1
enabled
disabled
disabled
disabled
disabled
1
WARNING
WARNING
NOTICE
0.0.0.0
0
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sample-interval
push-interval
boot-state
start-time
end-time
red-collect-state
red-max-trans
red-sync-start-time
red-sync-comp-time
push-success-trap-state
call-trace
internal-trace
log-filter
default-gateway
restart
exceptions
telnet-timeout
console-timeout
remote-control
cli-audit-trail
link-redundancy-state
source-routing
cli-more
terminal-height
debug-timeout
trap-event-lifetime
cleanup-time-of-day
last-modified-by
last-modified-date
task done
JC; Reviewed:
SPOC 11/8/2011
5
15
disabled
now
never
disabled
1000
5000
1000
disabled
disabled
disabled
all
192.168.3.11
enabled
0
0
enabled
enabled
disabled
enabled
disabled
24
0
0
00:00
admin@console
2008-09-24 17:09:04
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
53 of 54
Meg@POP-SM61
©2011
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 devconnect@avaya.com.
JC; Reviewed:
SPOC 11/8/2011
Solution & Interoperability Test Lab Application Notes
©2011 Avaya Inc. All Rights Reserved.
54 of 54
Meg@POP-SM61
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