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Alu Asm61 CS1K
Alu Asm61 CS1K
Abstract
Testing was conducted via the Internal Interoperability Program at the Avaya Solution
and Interoperability Test Lab.
Equipment Software/Firmware
Avaya Aura® Session Manager R6.1 SP3
Avaya S8510 Server
(Build 613006)
Avaya Aura® System Manager R6.1 SP3
Avaya S8800 Server
(Build 6.1.0.0.7345-6.1.5.112)
Avaya Communication Server 1000E R7.5,
Avaya Communication Server 1000E Version7.50.17
CP+PM Co-resident Linux Update 7.50_17 Sept13
Deplist: 7.50Q
Avaya 1140E Telephone UNIStim 0625C8A
Avaya 1120E Telephone UNIStim 0625C8A
Avaya 1120E Telephone SIP 4.01.15.00
Alcatel OmniPCX Enterprise Server 9.1 (I1.605-16-c)
Alcatel ipTouch NOE Telephone 4038 4.20.71
Alcatel ipTouch NOE Telephone 4068 4.20.71
Session Manager R6.1 provides all the SIP Proxy Service (SPS) and Network Connect
Services (NCS) functions previously provided by the Network Routing Server (NRS)
application. As a result, the Network Routing Server application is no longer needed to
configure a SIP trunk between Communication Server 1000E R7.5 and Session Manager
R6.1.
These instructions assume System Manager has already been configured as the Primary
Security Server for the Avaya Unified Communications Management application and
Communication Server 1000E is registered as a member of the System Manager Security
framework. For more information on how to configure System Manager to integrate with
the Avaya Unified Communications Management application, see Reference [5] in
Section 7.
In addition, these instructions also assume the configuration of the Call Server and SIP
Signaling Server applications has been completed and Communication Server 1000E is
configured to support the 1140e SIP & UNIStim telephones. For information on how to
administer these functions of Communication Server 1000E. See References [6] through
[8] in Section 7.
Using the System Manager web interface, the following administration steps will be
described:
Access Avaya Unified Communications Management (UCM) web interface from
System Manager
Confirm Node and IP addresses
Confirm Virtual D-Channel, Routes and Trunks
Configure SIP Trunk to Avaya Aura® Session Manager
Configure Route List Index and Distant Steering Code
Save Configuration
The Avaya Unified Communications Management Elements page will open in a new
browser window. Click on the Element Name corresponding to CS1000 in the Element
Type column.
The Node Details screen is displayed with additional details as shown below. Make a
note of the Call server IP address and Signaling Server TLAN IPv4 address fields
highlighted below as these values are used to configure other sections.
Expand Routes and Trunks on the left navigation panel and select Routes and Trunks (not
shown) to verify that a route with enough trunks to handle the expected number of
simultaneous calls has been configured.
Select Edit to verify the configuration. The details of the virtual Route defined for
sample configuration is shown below. Verify SIP (SIP) has been selected for Protocol
ID for the route (PCID) field and the Node ID of signaling server of this route
(NODE) and D channel number (DCH) fields match the values identified in the
previous section.
The values defined for the sample configuration are shown below.
The values defined for the sample configuration are shown below.
Scroll to the bottom of the page and click Save (not shown) to save SIP Gateway
configuration settings. Click Save on the Node Details screen (not shown). Select
Transfer Now on the Node Saved page as shown below.
Notes: The routing rule defined in this section is an example and was used in the sample
configuration. Other routing policies may be appropriate for different customer networks.
Under the Options section, select Route number of the route identified in Section 3.3.2
in the Route Number field and use default values for remaining fields as shown below.
Enter the following values and use default values for remaining fields.
Flexible Length number of digits Enter number of digits in dialed numbers
(In the sample configuration, 5 was used.)
Route List to be accessed for trunk steering code
Select Route List Index (Created in the
previous Section 3.5.1.)
In the main panel, a short procedure for configuring Network Routing Policy is shown.
The following screen shows the SIP Entity for Session Manager.
Under Port, click Add, and then edit the fields in the resulting new row.
Port Port number on which the system listens for SIP requests
Protocol Transport protocol to be used to send SIP requests
The following screen shows the Port definitions for the Session Manager SIP Entity.
Click Commit to save changes.
Click Commit to save changes. The following screen shows the Entity Links used in the
sample network.
In Session Manager, a default policy (24/7) is available that would allow routing to occur
at anytime. This time range was used in the sample network.
Click Commit to save changes. The following screen shows the Routing Policy Details
for calls to Communication Server 1000E.
A dial pattern must be defined that will direct calls to CS1000E. To configure the
CS1000E Dial Pattern select Dial Patterns on the left panel menu and then click on the
New button (not shown).
Under General:
Pattern Dialed number or prefix
Min Minimum length of dialed number
Max Maximum length of dialed number
Notes Comment on purpose of dial pattern
SIP Domain Select ALL
Use default values for the remaining fields. Click Commit to add this Session Manager.
Notes: All configuration is completed using the OXE manager menu. To enter the menu,
type mgr at the CLI prompt. Changes are committed with ctrl+v and cancel or return to
previous menu is ctrl+c.
This will show the license files installed on the system. Ensure that 185 and 188 are
greater than 0.
For the above values to hold true, all other options for compression in the Alcatel OXE
must be set to non-compressed options. Ensure the following parameters are set
accordingly:
Navigate to IP IP Domain
Intra-Domain Coding Algorithm default
Extra-Domain Coding Algorithm default
Navigate to IP TSC/IP
Default Voice Coding Algorithm without compression
Navigate to IP INT/IP
Default Voice Coding Algorithm without compression
Click on the SIP Entity Names Alcatel PBX and CS1K, shown in the previous screen,
and verify that the connection status is Up, as shown in the following screenshots. Alcatel
connection status is show below:
+==============================================================================+
| S I P T R U N K S T A T E Trunk group number : 10 |
| Trunk group name : To ASM60 |
| Number of Trunks : 62 |
+------------------------------------------------------------------------------+
| Index : 1 2 3 4 5 6 7 8 9 10 11 12 13 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 14 15 16 17 18 19 20 21 22 23 24 25 26 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 27 28 29 30 31 32 33 34 35 36 37 38 39 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 40 41 42 43 44 45 46 47 48 49 50 51 52 |
| State : F F F F F F F F F F F F F |
+------------------------------------------------------------------------------+
| Index : 53 54 55 56 57 58 59 60 61 62 |
| State : F F F F F F F F F F |
+------------------------------------------------------------------------------+
| F: Free | B: Busy | Ct: busy Comp trunk | Cl: busy Comp link |
| WB: Busy Without B Channel| Cr: busy Comp trunk for RLIO inter-ACT link |
| WBD: Data Transparency without chan.| WBM: Modem transparency without chan. |
| D: Data Transparency | M: Modem transparency |
+------------------------------------------------------------------------------+
7. Additional References
Product documentation for Avaya products may be found at http://support.avaya.com
[1] Avaya Aura® Session Manager Overview, Doc # 03603323, Issue 1 Release 6.1
[2] Administering Avaya Aura® Session Manager, Doc # 03603324, Issue 1 Release
6.1
[3] Maintaining and Troubleshooting Avaya Aura® Session Manager, Doc #
03603325, Issue 1 Release 6.1
[4] IP Peer Networking Installation and Commissioning, Release 7.5, Document
Number NN43001-313, available at http://support.avaya.com
[5] Unified Communications Management Common Services Fundamentals, Avaya
Communication Server 1000E Release 7.5, Document Number NN43001-116,
available at http://support.avaya.com
[6] SIP Line Fundamentals, Release 7.5, Document Number NN43001-508, Issue
02.03, available at http://support.avaya.com
[7] Co-resident Call Server and Signaling Server Fundamentals, Avaya
Communication Server 1000E Release 7.5, Document Number NN43001-509,
available at http://support.avaya.com
[8] Signaling Server and IP Line Fundamentals, Avaya Communication Server
1000E Release 7.5, Document Number NN43001-125, available at
http://support.avaya.com
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 Solution & Interoperability Test Lab at interoplabnotes@list.avaya.com