Application Notes for Biamp Tesira SVC

Application Notes for Biamp Tesira SVC
Avaya Solution & Interoperability Test Lab
Application Notes for Biamp Tesira SVC-2 and Avaya
Communication Server 1000 SIP Line Release 7.5 – Issue 1.0
Abstract
These Application Notes describe a solution comprised of Avaya Communication Server 1000
SIP Line Release 7.5 and Biamp Tesira SVC-2. The overall objective of the interoperability
compliance testing is to verify Biamp Tesira SVC-2 functionalities in an environment
comprised of Avaya Communication Server 1000 SIP Line with various Avaya Unistim and
SIP IP Telephones.
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.
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1. Introduction
These Application Notes describe the procedures for configuring Biamp Tesira SVC-2 which
was compliance tested with Avaya Communication Server 1000 SIP Line Release 7.5.
The Tesira SVC-2 enable conferencing over VoIP directly from Tesira SERVER-IO, with two
channels of VoIP interface per card. Tesira SVC-2 allows Tesira SERVER-IO to connect directly
to IP-based phone systems and eliminate the need for VoIP adapters. Used in conjunction with
SEC-4 4-Channel Wideband Acoustic Echo Cancellation Input Cards and STC-2 Dual-Channel
Telephone Interface Cards, the Tesira SVC-2 makes the Tesira SERVER-IO telephone
conferencing product powerful, flexible, and affordable. Combined with the STC-2 Card, the
Tesira SVC-2 makes it possible to create redundancies within a conferencing system for multipoint conferences and/or back-up to VoIP lines. Up to 6 Tesira SVC-2 can be installed into a
single Tesira SERVER-IO unit.
2. General Test Approach and Test Results
All test cases were performed manually. The general approach was to place various types of calls
to and from Biamp Tesira SVC-2. Biamp Tesira SVC-2 operations such as inbound calls,
outbound calls, hold, and Biamp Tesira SVC-2 interactions with CS 1000 SIP Line and Avaya
SIP and Unisitm telephones were verified.
DevConnect Compliance Testing is conducted jointly by Avaya and DevConnect members. The
jointly-defined test plan focuses on exercising APIs and standards-based interfaces pertinent to
the interoperability of the tested products and their functionalities. DevConnect Compliance
Testing is not intended to substitute a full product performance or feature testing performed by
third party vendors, nor is it to be construed as an endorsement by Avaya of the suitability or
completeness of a third party solution.
2.1. Interoperability Compliance Testing
The focus of this testing was to verify that the Biamp Tesira SVC-2 was able to interoperate with
the CS 1000 SIP Line system. The following areas were tested:
 Registration of Biamp Tesira SVC-2 Lines to the CS1000 SIP Line Gateway.
 Call establishment of Biamp Tesira SCV-2 with CS1000 SIP and Unistim telephones.
 Telephony features: basic calls, conference, DTMF (dual tone multi frequency)
RFC2833, SIP INFO and INBAND transmission, voicemail, busy, hold, call waiting.
 PSTN calls over PRI trunk.
 Codec negotiation – G.711 and G.729.
2.2. Test Results
The objectives outlined in the Section 2.1 were verified. The following observations were made
during the compliance testing:
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


Biamp Tesira SVC-2 allows configuration to send both DTMF tones outband as per
RFC2833 and DTMF tones inband to the CallPilot messaging system simultaneously.
This can cause the CallPilot not to recognize inputted digits correctly. Biamp Tesira
SVC-2 needs to be configured to send only one type of DTMF tones.
Biamp Tesira SVC-2 supports UDP, TCP and TLS transport protocols but TCP doesn’t
work with CS 1000 SIP Line server. It is recommended to use UDP protocol to work
with CS 1000 SIP Line.
Local conference made by Biamp Tesira SVC-2 has a choppy and noisy audio if different
party uses different codec. All parties need to use the same codec to ensure audio quality.
2.3. Support
Technical support for Biamp Tesira SVC-2 can be obtained by contacting Biamp at:
 http://www.biamp.com/support/index.aspx
 (800)-826-1457
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3. Reference Configuration
Figure 1 illustrates the test configuration used during the compliance testing between the Avaya
CS1000 SIP Line Release 7.5 and the Biamp Tesira SVC-2.
Figure 1: Network Configuration Diagram
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4. Equipment and Software Validated
The following equipment and software was used during the lab testing:
Equipment / Software
Avaya CS1000E
Avaya CS 1000 SIP Line
Avaya CallPilot® Messaging System
Avaya IP Phone 1140
Avaya IP Phone 2004P2
Avaya SIP 1140
Biamp Tesira SVC-2
Biamp Tesira
Biamp Linux
Release / Version
Call Server (CPPM): 7.50Q
Signaling Server (CPPM): 7.50.17
7.50.17
5.0.1
0625C6O
0692D93
02.02.21.00
1.0.0.37
1.0.0.37
2.6.32.28-Biamp
5. Configure Avaya CS 1000 - SIP LINE
This section describes the steps to configure the Avaya CS1000 SIP Line using CS 1000 Element
Manager. A command line interface (CLI) option is available to provision the SIP Line
application on the CS 1000 system. For detailed information on how to configure and administer
the CS 1000 SIP Line, please refer to Section 9 [1].
The following is the summary of tasks for configuring the CS 1000 SIP Line:
 Log in to Unified Communications Management (UCM) and Element Manager (EM).
 Enable SIP Line Service.in Customer Data Block.
 Create SIP Line Telephony Node.
 Create D-Channel for SIP Line.
 Create an Application Module Link (AML).
 Create a Value Added Server (VAS).
 Create a Virtual Trunk Zone.
 Create a Route Data Block (RDB).
 Create SIP Line Virtual Trunks.
 Create SIP Line phones.
5.1. Prerequisite
This document assumes that the CS1000 SIP Line server has been:
 Installed with CS 1000 Release 7.5 Linux Base.
 Joined CS 1000 Release 7.5 Security Domain.
 Deployed with SIP Line Application.
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The following packages need to be enabled in the key code. If any of these features have not
been enabled, please contact your Avaya account team or Avaya technical support at
http://support.avaya.com.
Package Mnemonic
SIP_LINES
FFC
SIPL_AVAYA
SIPL_3RDPARTY
Package #
Descriptions
Package Type
Applicable market
417
SIP Line Service
New package
Global
package
139
Flexible Feature Codes
Existing package
Global
415
Avaya SIP Line
Existing package
Global
Existing package
Global
package
416
Third-Party SIP Line
Package
5.2. Log in to Unified Communications Management (UCM) and
Element Manager (EM)
Use the Microsoft Internet Explorer browser to launch CS 1000 UCM web portal at http://<IP
Address or FQDN> where <IP address or FQDN> is the UCM Framework IP address or FQDN
for UCM server.
Log in with the username/password which was defined during the primary security server
configuration, the UCM home page appears as shown in the Figure 2 below.
Figure 2: The UCM Home Page of CS 1000 Release 7.5
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On the UCM home page, under the Element Name column, click on the EM name of CS 1000
system that needs to be configured, in this sample that is cpppm3 (see Figure 2). The CS 1000
Element Manager page appears as shown in Figure 3 below.
Figure 3: CS 1000 Release 7.5 EM Home Page
5.3. Enable SIP Line Service in Customer Data Block
On the EM page, navigate to Customers on the left column menu; select the customer number to
be enabled with SIP Line Service (not shown).
 Enable SIP Line Service by clicking on the SIP Line Service check box.
 Enter the prefix number in the User agent DN prefix text box as shown in Figure 4.
Figure 4: SIP Line Service in Customers Data Block
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5.4. Create SIP Line Telephony Node
On the EM page, navigate to menu System  IP Network  Nodes: Servers, Media Cards.
Click Add to add a new SIP Line Node to the IP Telephony Nodes. The new IP Telephony Node
page appears as shown in Figure 5
Enter the information as shown below:
 Node ID: e.g., 512. This is the node ID of SIP Line server.
 Call Server IP Address: e.g. 10.10.97.78.
 Node IP Address: e.g. 10.10.97.187. This is the SIP Proxy IP address that SIP endpoint
uses to register to.
 Subnet Mask: 255.255.255.192.
 Embedded LAN (ELAN) Gateway IP Address: e.g. 10.10.97.65.
 Embedded LAN (ELAN) Subnet Mask: 255.255.255.192.
 Check SIP Line check box to enable SIP Line for this Node.
Figure 5: Adding a New IP Telephony Node
Click on the Next button to go to next page.
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The page, New IP Telephony Node with Node ID, will appear as shown in Figure 6:
- Click Select to Add drop-down menu list to select the desired server to add to the node.
- Click Add button
- Select the check box next to the newly added server, and click Make Leader (not
shown).
Figure 6: Adding a New IP Telephony Node (cont)
Click on the Next button to go to next page.
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The SIP Line Configuration Detail page appears as shown in Figure 7:
 Enter SIP Line domain name in SIP Domain name text box, e.g., sipl75.com.
Note: SIP domain name can be either a domain name or IP address, in this sample the
SIP domain name is presented.
 Check the Enable gateway service on this node option for SIP Line Gateway
Application.
Figure 7: Adding a new IP Telephony Node (cont)
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Scroll down to the Branch / GR Office Settings section under SIP Line Gateway Service as
shown in Figure 8:
 Select MO from the SLG Role list.
 From the SLG Mode list, select S1/S2 (SIP Proxy Server 1 and Server 2).
Figure 8: Adding a new IP Telephony Node (cont)
Click Next. The Confirm new Node details page appears (not shown).
Click on the Transfer Now button, the Synchronize Configuration Files (Node ID 512) page
appears (not shown). Click Finish and wait for the configuration to be saved.
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The Node Saved page appears, as shown in Figure 9. Click the Transfer Now button.
Figure 9: Node Saved with Transfer Configuration
Select the SIP Line server associated with changes and then click on the Start Sync button to
transfer the configuration files to the selected servers as shown in Figure 10.
Figure 10: Synchronize Configuration Files
Note: The first time a new Telephony Node is added and transferred to the call server, the SIP
Line services need to be restarted. To restart the SIP Line services, log in as administrator to the
command line interface of the SIP Line server and issue command: appstart restart.
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5.5. Create D-Channel for SIP Line
On the EM page, navigate to Routes and Trunks  D-Channels.
Under the Configuration section as shown in Figure 11, enter a number in the Choose a DChannel Number field, and click on the to Add button.
Figure 11: D-Channels configuration page
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The D-Channels xx Property Configuration page appears as shown in Figure 12 where xx is
the D-Channel number to add.
- From the Interface type for D-channel (IFC) list, select Meridian Meridian1 (SL1).
- Leave the other fields at default values.
Figure 12: SIP Line D-Channel Property Configuration
Click on the Basic options (BSCOPT) link. The Basic options (BSCOPT) list expands (not
shown). Click on Edit to configure Remote Capabilities (RCAP).
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The Remote Capabilities Configuration Details page will appear as shown in Figure 13.
 Select the Message waiting interworking with DMS-100 (MWI) check box. This
option must be enabled to support voice mail notification on SIP Line endpoints.
 Select the Network name display method 2 (ND2) check box. This option must be
enabled to support name display between SIP Line endpoints.
 Other check boxes are left unchecked.
 At the bottom of the Remote Capabilities Configuration Details page, click Return Remote Capabilities to return to the D-Channel xx Property Configuration page.
Figure 13: SIP Line D-Channel RCAP Configuration Details
Click on the Submit button of the D-Channel Property Configuration page to save changes.
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5.6. Create an Application Module Link (AML)
On the EM page, navigate to System  Interfaces  Application Module Link, click on the
Add button to add a new Application Module Link (not shown). The New Application Module
Link page appears as shown in Figure 14.
Enter an AML port number in the Port number text box. The AML of SIP Line Service can use
a port from 32 to 127. In this case, SIP Line Service is configured to use port 33.
Click on the Save button to complete adding the AML link, and to save the configuration.
Figure 14: Adding a new Application Module Link
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5.7. Create a Value Added Server (VAS)
On the EM page, navigate to System  Interfaces  Value Added Server and click on the
Add button to add a new VAS.
The Value Added Server page appears (not shown). On this page, select the Ethernet Link and
the Ethernet Link page appears as shown in Figure 15.
Enter a number in the Value added server ID field, in this example 33 was used. In the
Ethernet LAN Link drop-down list, select the AML number of ELAN that was created in
Section 5.6.
Leave other fields at default values and click on the Save button to complete adding the VAS
and save the configuration.
Figure 15: Adding a new Value Added Service for the AML
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5.8. Create a Virtual Trunk Zone
On the EM page, navigate to menu System  IP Network  Zones. The Zones page appears
on the right (not shown). On this page select Bandwidth Zones link.
On the Bandwidth Zones page (not shown), click on the Add button, the Zone Basic Property
and Bandwidth Management page appears as shown in Figure 16.
Enter a zone number in the Zone Number (Zone) field. In the Zone Intent (ZBRN) drop-down
menu select VTRK (VTRK).Leave other fields at default values and click on the Save button to
complete adding the Zone.
Note: Repeat the step above to create another zone for the SIP Line phone; however remember to
select MO (MO), instead of VTRK in the field Zone Intent.
Figure 16: Adding a new Zone for Virtual Trunk
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5.9. Create a Route Data Block (RDB)
On the EM page, navigate to the menu Routes and Trunks  Routes and Trunks; the Routes
and Trunks page appears (not shown). On this page, click on the Add route button next to the
customer number that the route will belong to.
The Customer ID, New Route Configuration page appears, expand the Basic Configuration
tab, and enter values as shown in Figure 17 and 18.
 Route Number (ROUT): 3
 Trunk type(TKTP): TIE
 Incoming and Outgoing trunk (ICOG): Incoming and Outgoing (IAO)
 Access code for the trunk route (ACOD): a number for ACOD, for example 8003.
 The route is for a virtual trunk route (VTRK): checked.
 Zone for codec selection and bandwidth management (ZONE): 4, this is the Virtual
trunk zone number created in the Section 5.8.
 Node ID of signaling server of this route (NODE): 512, this is the node ID of the SIP
Line Telephony Node created in Section 5.4.
 Protocol ID for the route (PCID): SIP Line (SIPL).
 Integrated services digital network option (ISDN): checked.
 Mode of operation (MODE): Route uses ISDN Signaling Link (ISLD).
 D channel number (DCH): 4, the D-channel number that was created in the Section 5.5.
 Interface type for route (IFC): Meridian M1 (SL1).
 Network calling name allowed (NCNA): checked.
 Channel type (CHTY): B-channel (BCH).
 Call type for outgoing direct dialed TIE route (CTYP): Coordinate dialing plan
(CDP).
 Calling Number dialing plan (CNDP): Coordinate dialing plan (CDP).
Leave default values for The Basic Route Options, Network Options, General Options, and
Advanced Configurations sections.
Click the Submit button to complete adding the route and save configuration.
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Figure 117: SIP Line Route Basic Configuration
Figure 18: SIP Line Route Basic Configuration (cont)
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5.10. Create SIP Line Virtual Trunks
On the EM page, navigate to Routes and Trunks  Routes and Trunks and select the Add
trunk button beside the route created in the Section 5.9 to create new trunks.
The Customer ID, Route ID, Trunk type TIE trunk data block page appears as shown in
Figure 19, enter values for fields as shown below:
 Multiple trunk input number: 32 for creating 32 trunks.
 Auto increment member number: checked.
 Trunk data block: IP Trunk (IPTI).
 Terminal Number (TN): 100 0 2 0, the first TN of a TN range.
 Member number: 33, ID for first trunk; next ID’s will be automatically created.
 Start arrangement Incoming: Immediate (IMM).
 Start arrangement Outgoing: Immediate (IMM).
 Trunk Group Access Restriction: 1.
 Channel ID for this trunk: 33, this ID should be the same as ID of Member number.
Click on the Edit button for Class of Service and assign following class of services (not shown):
- Media security: Media Security Never (MSNV).
- Restriction level: Unrestricted.
Leave other fields at default values and click on the Return Class of Service button to return to
the Customer ID, Route ID, Trunk type TIE trunk data block page, then click on the Save
button to complete adding virtual trunks for SIP Line.
Figure 19: Adding virtual trunks for SIP Line Trunk
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5.11. Create SIP Line Phones
To create a SIP Line phone on the Call Server, log in as administrator and use overlay command
LD 20 as shown below.
The bold fields must be properly inputted as they are configured on the Call server, for other
fields press <Enter> to leave it at default values.
Figure 20: Sample of creating a new SIP Line Phone
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6. Configure Biamp Tesira SVC-2
Biamp installs, configures, and customizes the Tesira SVC-2 application for their end customers.
How to configure a Tesira system is beyond the scope of these application notes. This section
only provides steps to configure Biamp Tesira SVC-2 to interface with Avaya CS 1000 SIP Line
server. For more information on how to administer Biamp Tesira SVC-2 please refer to the
Tesira SVC-2 documents from Biamp .
Select the Tesira icon from Workstation where Tesira software was installed to start Tesira
software.

Highlight the VoIPControl/Status block, as shown below.
Figure 21: The main window of Tesira Software
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
Righ-click mouse button and select Properties, the Properties menu will display on the
right
Figure 22: Properties window of VoIP Control
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
Form the Properties window, select DSP Properties tab and navigate to Protocol
SIPTransport to configure transport to be used. Select UDP transport (default
setting).
Figure 23: Properties window with UDP Transport
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
Select Line Properties under the General section
Figure 24: Line Properties
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
From the VoIP Line Properties window, click the Protocol tab.
Figure 25: VoIP Line Properties Window
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
In the Protocol tab, provide the following information:
o SIP User Name – Enter a user created in Section 5.11.
o Authentication User Name – Enter a user created in Section 5.11.
o SIP Domain Name – Enter the SIP domain name configured in Section 5.4
o Authentication Password – Enter the password for the user in Section 5.11
o Proxy Vendor – Select Avaya CS 1000.
o Proxy Address – Enter the IP address of CS 1000 SIP Line server.
o Proxy Port – Enter either 5060 or 5061.
 TLS – 5061
 UDP – 5060
o Click on the OK button. Default values may be used for all other fields.
Note: Biamp Tesira SVC-2 can provide two inbound extensions (L1 and L2).
Figure 26: Protocol tab of VoIP Line Properties
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Note: in case the SIP Line domain name in Section 5.4 is configured to use IP address
instead of SIP domain name, the registration of Biamp Tesira SIP user is configured as in
Figure 28 below. The SIP Domain Name field is left as blank.
Figure 27: Protocol tab of VoIP Line Properties
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7. Verification Steps
This section includes some steps that can be followed to verify the configuration.
 Verify that the Biamp Tesira SVC-2 registers successfully with the CS 1000 SIP Line
Gateway server and Call Server by using the CS 1000 Linux command line and CS 1000
Call Server overlay LD 32.
- Log in to the SIP Line server as an administrator by using Avaya account.
- Issue command “slgSetShowByUID [userID]” where [userID] is SIP Line user’s
ID being checked.
[admin@sipl ~]$ slgSetShowByUID 54350
=== VTRK ===
UserID
AuthId
TN
Clients Calls
SetHandle Pos ID
SIPL Type
--------------- ---------- --------------- ------- ---------- --------------54350
54350
104-00-00-10
1
0
0x8fc4cf8
SIP Lines
StatusFlags = Registered Controlled KeyMapDwld SSD
FeatureMask =
CallProcStatus = 0
---
Current Client = 0, Total Clients = 1
== Client 0 ==
IPv4:Port:Trans
Type
=
UserAgent
=
x-nt-guid
=
RegDescrip
=
RegStatus
=
PbxReason
=
SipCode
=
hTransc
=
Expire
=
Nonce
=
NonceCount
=
hTimer
=
TimeRemain
=
Stale
=
Outbound
=
ClientGUID
=
MSec CLS
=
Contact
=
KeyNum
=
AutoAnswer
=
Key
0
1
2
3
4
5
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Func
3
126
9
29
22
2
Lamp
0
0
0
0
0
0
= 10.10.98.42:5060:udp
SIP3
Tesira/1.0.0.37
267d228547c1562399f1f743a2971fb5
1
OK
200
(nil)
3600
f56a9946ba497bde7eb445efb518f4f1
2
0x8f64e60
1338
0
0
0
MSNV (MSEC-Never)
sip:54530@10.10.98.42:5060
255
NO
Label
54009
2654009
54334
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17
18
19
20
21
22
24
25
26
16
18
27
19
52
25
11
30
31
0
0
0
0
0
0
0
0
0
== Subscription Info ==
Subscription Event = None
Subscription Handle = (nil)
SubscribeFlag = 0
-
Log in to the call server using the admin account.
Load overlay 32 and then issue command “stat [TN]” where [TN] is the SIP Line
user’s TN being checked
>ld 32
NPR000
.stat 104 0 0 10
IDLE REGISTERED 00


Place a call from and to Biamp Tesira SVC-2 SIP user and verify that the call is
established with 2-way speech path.
During the call, use a sniffer tool (ethereal/wireshark) at the SIP Line Gateway and
clients to make sure that all SIP request/response messages are correct.
8. Conclusion
All of the executed test cases have passed and met the objectives outlined in tSection 2.1, with
some exceptions outlined in Section 2.2. The Biamp Tesira SVC-2 is considered to be in
compliance with Avaya CS 1000 SIP Line System Release 7.5.
9. Additional References
Product documentation for the Avaya CS 1000 products may be found at:
https://support.avaya.com/css/Products/
Avaya CS1000 Documents:
[1] Avaya Communication Server 1000E Installation and Commissioning.
[2] Avaya Communication Server 1000 SIP Line Fundamental, Release 7.5.
[3] Avaya Communication Server 1000 Element Manager System Reference –
Administration.
[4] Avaya Communication Sever 1000 Co-resident Call Server and Signaling Server
Fundamentals.
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Solution & Interoperability Test Lab Application Notes
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[5] Avaya Communication Server 1000 Unified Communications Management Common
Services Fundamentals.
[6] Avaya Communication Server 1000 ISDN Primary Rate Interface Installation and
Commissioning.
Product documentation for the Biamp Tesira SVC-2 products may be found at:
http://www.biamp.com
KP; Reviewed:
SPOC 4/17/2012
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
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©2012 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.
KP; Reviewed:
SPOC 4/17/2012
Solution & Interoperability Test Lab Application Notes
©2012 Avaya Inc. All Rights Reserved.
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