Avaya SCS Solutions Test Lab

SCS 4.0 SIP Trunking Configuration Guide for Skype Connect

Issue 1.0

Abstract

This document provides guidelines for deploying SCS 4.0 with Skype using SIP Trunking.

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Table of Contents
1. INTRODUCTION ..................................................................................................................................................................... 3
1.1. DOCUMENT CHANGE HISTORY ..................................................................................................................................................... 3

2. SCS SOFTWARE VERSIONS ................................................................................................................................................. 4

3. SUPPORTED FEATURES ....................................................................................................................................................... 5

3.1. TECHNICAL CAVEATS ................................................................................................................................................................... 5

4. OVERVIEW .............................................................................................................................................................................. 6
4.1. NETWORK DIAGRAM .................................................................................................................................................................... 6

5. CONFIGURATION GUIDE .................................................................................................................................................... 7

5.1. REQUIREMENTS ............................................................................................................................................................................ 7
5.2. SCS CONFIGURATION ................................................................................................................................................................... 7

6. REFERENCES ........................................................................................................................................................................ 10

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1. Introduction
The goal of this document is to provide a reference to Avaya Authorized Partners for configuring a Avaya
SCS 4.0 to interoperate with SIP for Skype.

From reading this document the reader should gather a good understanding of capabilities, limitations
and configuration for this solution.

This document does not describe procedures to configure the SCS for advanced functionality. For more
information and procedures, please refer to the Avaya technical documentation found on the Avaya
website.

1.1. Document Change History

Draft 0.1 December 16, original version

Issue 1.0 January 10, issued

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2. SCS Software Versions
SCS software versions should be at the following levels: The SCS Software version mentioned here
should be the same as the release candidate. Please check.
• SCS Base Software - 4.0.4-009233017289 or higher
• Nortel 1200 Sets – SIP12x0.01.01.01.00 or higher
• Polycom Sets – 3.1.3 rev. B or higher
• Nortel SMC 3456 Client – Build – 53737 or higher

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3. Supported Features
The following are capabilities provided by this solution:
• Basic calls (G711a-law 20 ms, G729 20ms)
• Calling line (number) identification presentation (CLIP)
• DTMF (RFC2833)
• Call hold
• Call transfer (Blind and consultative transfers)
• Ad hoc conference calls
• Call forward
• Call Redirection to Voice Mail on SCS
• Incoming calls from Skype soft client.

3.1. Technical Caveats

The following are limitations with this solution:

• Skype does not support Early Media in SIP.

• When an incoming call to the SCS is unanswered, the SCS sends a 486 busy here.
The SKYPE client sees this as being a busy number and displays two options. Option
1 is to start a redial process and Option 2 is to cancel the call.
• When an incoming call is made to an unknown number on the SCS, the SCS sends
reply 407. The SKYPE client sees this as being a busy number and displays two
options. Option 1 is to start a redial process and Option 2 is to cancel the call.
• An incoming external call forwarded or transferred to an external number fails
because the SIP INVITE has the original callers calling information. Skype does not
accept this and sends a 403 forbidden.
• An incoming external call to the SCS extension transfers but disconnects before the
call is answered by another external party. The SIP invite going to SKYPE has the
FROM header incorrect.

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4. Overview
4.1. Network Diagram
The follow represents a network diagram of SCS connected to Skype via a SIP trunk:

Figure 4-1 SIP Trunk between SCS and Skype

Skype soft client

Public Internet

NAT/FW

SCS

Skype Soft

PSTN

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5. Configuration Guide

5.1. Requirements
The following are additional minimum requirements that must be met to configure Skype Connect
between SCS and Skype:
• Skype Connect Account(s) or SIP profile with User ID and password
• SCS must have SIP Trunking Server Role added as outlined in SCS System Configuration task
Based Guide for SCS 4.0.
• Customers that are behind NAT must Configure NAT Traversal as detailed in SCS 4.0 Device
Configuration – Gateways Task Based Guide.

5.2. SCS Configuration

To Add a SIP Trunking role:
1. Navigate to System > Servers > server name > Configure
2. Here, under 'Server Roles' select the checkbox for 'SIP Trunking'
3. Click on OK button
4. You will be prompted to restart a number of services on the SCS. Click the here link.
5. Select the affected services and click the Restart button
6. sipXbridge-1 now gets added on the Devices > SBCs screen

To configure a SIP trunk between SCS and Skype, do the following:

7. In the SCS home page, select Devices followed by Gateways.

8. The Gateways page will be displayed.
9. From the Add New Gateway drop down box, select SIP Trunk.
10. The SIP Trunk configuration page will be displayed.
11. In the Name field, provide a descriptive name for the SIP trunk to Skype.
12. In the Route drop down box, select sipXbridge-1
13. From the User Provider Template drop down box, select Skype for SIP
14. The Address field will be automatically populated with sip.skype.com
15. Click OK button
16. The Gateways page will be displayed
17. Click on the name of the Gateway created above
18. The SIP Trunk configuration page for the gateway will be displayed.
Note: Skype currently requires the From header in each outbound call to be equal to the Skype
account user ID. Else the outbound calls will fail. Hence it is mandatory to configure the Caller ID
in the Skype SIP Trunk.
19. On the left hand navigation pane, click on Caller ID.
20. Click on the Show Advanced Settings link

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 21. Select the checkbox for Specify Caller ID and click anywhere on the screen
22. In the Caller ID field below Specify Caller ID type your SIP User name (username provided by
Skype in the BCP under Skype for SIP and SIP profile settings) This is typically a 14 digit numeric
number starting with 99050
23. Click on the Apply button
24. Click on ITSP Account on the left hand navigation pane
25. The ITSP page will be displayed. The ITSP server domain name field will be
populated with the details for Skype.
26. In the User Name field, enter the SIP User Name provided by Skype. Again, this
is a 14 digit numeric number
27. In the Password field, enter the password for the above user that was provided by Skype in the
SIP profile settings
28. Click the Apply button.
29. You will be prompted to restart a number of services on SCS. Click the Here, link.
30. Select the affected services and click the Restart button

31. A Dial Plan can now be configured to provide SCS sets with access to PSTN via Skype for SIP
From the System menu, select Dial Plans.
32. The Dial Plans page will be displayed.
33. In the Add New Rule drop down list, select Custom to create a custom dial plan
34. The Dial Rule page will be displayed.
35. Click on the Enabled checkbox.
36. Enter a Name and Description for the rule
37. In the Prefix field enter the digit user’s will dial in order to utilize this Dial Rule. Any number of
digits has also been selected, as this will provide for varying lengths of number strings to be
dialed
38. In the Resulting Call fields, select the Matched Suffix dropped the prefix before the remaining
digits are forwarded to Skype.
39. The Skype SIP trunk can now be selected from the Gateway More Actions drop down box. This
is the Skype for SIP settings configured previously.
40. The Gateway will be displayed. Click the Apply button followed by the OK button.
41. The Dial Plan rule will be displayed. Restart any services as prompted.

1
To terminate inbound calls from Skype on the Auto Attendant on SCS , do the following

42. From the System menu, select Dial Plans.

43. The Dial Plans page will be displayed.

1
This explains how to route calls to the SCS auto attendant. If you require the calls to route elsewhere
such as to any SIP user, a Hunt Group or an ACD line, please reference the SCS configuration
documents.

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 44. Click on the Auto Attendant dial plan.
45. In the Auto Attendant aliases text box add your SIP for Skype user ID. This is typically a 14 digit
numeric number.
46. Click OK button.
47. The Dial Plan rule will be displayed. Restart any services as prompted.

If the SCS is behind a NAT and you want SCS to perform local NAT compensation, do the following:
48. Navigate to System > Internet Calling > NAT Traversal
49. Here, select the checkbox for Server Behind NAT if your system is behind NAT
50. Click on Apply
51. Restart any services if prompted

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6. References
The following are useful references to assist in this solution:
1. SCS 4.0 Task Based Guides
http://www.nortelscs.com/scs/sites/default/files/scsr3_docs/home.html
2. SCS 4.0 Online Help (available in the SCS Web UI directly). Can be viewed here also:
http://scsavaya.com/scs/sites/default/files/scs3help/start.htm

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 3.

©2010 Avaya Inc. All Rights Reserved.

Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and
™ are registered trademarks or trademarks, respectively, of Avaya Inc. Nortel, Nortel
Networks, the Nortel logo, and the Globemark are trademarks of Nortel Networks. 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.

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