SCS Configuration Guide for Skype for SIP. Issue 1.0

Transcription

1 SCS Configuration Guide for Skype for SIP Issue 1.0 Abstract This document provides guidelines for deploying SCS 3.2 with Skype using SIP Trunking. 1 of Avaya Inc. All Rights Reserved. SCS- Skype Config.doc Page 1 of 11

2 Table of Contents 1. INTRODUCTION DOCUMENT CHANGE HISTORY SCS SOFTWARE VERSIONS SUPPORTED FEATURES TECHNICAL CAVEATS OVERVIEW NETWORK DIAGRAM CONFIGURATION GUIDE REQUIREMENTS SCS CONFIGURATION REFERENCES of Avaya Inc. All Rights Reserved. SCS- Skype Config.doc Page 2 of 11

3 1. Introduction The goal of this document is to provide a reference to Avaya Authorized Partners for configuring a Avaya SCS 3.0 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 Document Change History Draft 0.01 October 5, 2009, original version Draft 0.4 Addressed various comments Issue 1.0 March 10, of 11

4 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 or higher Nortel 1200 Sets SIP12x or higher Polycom Sets rev. B or higher Nortel SMC 3456 Client Build or higher 4 of 11

5 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 Technical Caveats The following are limitations with this solution: Hairpinned consultative transfer of Inbound /Outbound calls fails using SMC softphone (when Skype is the ITSP) Low probability call flow of transfer out of Skype for SIP of inbound Skype calls only with SMC phone. Inbound call does not overflow to a remote worker after a maximum wait time(if ITSP is skype) Low probability call flow of remote work overflow ACD. Hairpinned Blind transfer of inbound / outbound calls fails (when MOH is enabled on both sipxbridge and phones, ITSP is Skype) SCS 3.0 does not support MOH on both sipxbridge and phones together. An inbound call from Skype terminating in a paging group has no audio flow Not a supported call flow Voic caller identification uses "Outside caller" when callers user address has a leading '+' Existing issue with all ITSP s. For incoming ITSP / sipxbridge call no Ringback tone is heard by external caller after a Blind Transfer Existing issue with all ITSP s. 5 of 11

6 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 Public Internet Skype soft client NAT/FW SCS M Skype Soft PSTN 6 of 11

7 5. Configuration Guide 5.1. Requirements The following are additional minimum requirements that must be met to configure Skype for SIP between SCS and Skype: Skype for SIP 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 NN Issue 1.1 SCS 3.0 Page 98 Customers that are behind NAT must Configure NAT Traversal as detailed in SCS 3.0 Device Configuration Gateways Task Based Guide NN Issue 1.3 Page 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. From the User Provider Template drop down box, select Skype for SIP 13. The Address field will be automatically populated with sip.skype.com 14. In the Route drop down box, select sipxbridge 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 7 of 11

8 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 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. To terminate inbound calls from Skype on the Auto Attendant on SCS 1, 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. 8 of 11

9 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 9 of 11

10 6. References The following are useful references to assist in this solution: 1. SCS 3.0 Task Based Guides 2. SCS 3.0 Online Help (available in the SCS Web UI directly). Can be viewed here also: 10 of 11

11 3. 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. 11 of 11