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Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
Dial Plan
Dial Plan. Enter the dial plan script for this line. Refer to “Appendix C: Dial Plan and Auto-Attendant Scripting for
Advanced Users” for more details.
Streaming Audio Server (SAS)
SAS Enable. To enable the use of the line as a streaming audio source, select yes. Otherwise, select no. If
enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming calls and streams audio
RTP packets to the caller. The default is no.
SAS DLG Refresh Intvl. If this is not zero, it is the interval at which the streaming audio server sends out session
refresh (SIP re-INVITE) messages to determine if the connection to the caller is still active. If the caller does not
respond to the refresh message, then the System will end this call with a SIP BYE message. The range is 0 to
255 seconds (0 means that the session refresh is disabled).The default is 30.
SAS Inbound RTP Sink. This setting works around devices that do not play inbound RTP if the streaming audio
server line declares itself as a send-only device and tells the client not to stream out audio. Enter a Fully Qualified
Domain Name (FQDN) or IP address of an RTP sink; this will be used by the System’s streaming audio server line
in the SDP of its 200 response to an inbound INVITE message from a client.
Call Feature Settings
Blind Attn-Xfer Enable. This settings lets the System perform an attended transfer operation by ending the
current call leg and performing a blind transfer of the other call leg. If this feature is disabled, the System
performs an attended transfer operation by referring the other call leg to the current call leg while maintaining
both call legs. To use this feature, select yes. Otherwise, select no. The default is no.
MOH Server. Enter the user ID or URL of the auto-answering streaming audio server. When only a user ID is
specified, the current or outbound proxy will be contacted. Music-on-hold is disabled if the MOH Server is not
specified.
Xfer When Hangup Conf. This setting makes the System perform a transfer when a conference call has ended.
Select yes or no from the drop-down menu. The default is yes.
Conference Bridge URL. This feature supports external conference bridging for n-way conference calls (n > 2),
instead of mixing audio locally. To use this feature, set this parameter to that of the server's name, e.g.,
conf@myserver.com:12345 or conf (which uses the Proxy value as the domain).
Conference Bridge Ports. Select the maximum number of conference call participants. The range is 3 to 10. The
default is 3.
Figure 6-41: Voice - FXS 1 Screen - Call Feature Settings
Figure 6-39: Voice - FXS 1 Screen - Dial Plan
Figure 6-40: Voice - FXS 1 Screen - Streaming Audio
Server
VoIPon Solutions www.voipon.co.uk [email protected] Tel: +44 (0) 1245 600560
- IP Telephony 1
- Copyright and Trademarks 2
- How to Use this Guide 2
- Table of Contents 3
- List of Figures 5
- Chapter 1: Introduction 8
- What’s in this Guide? 9
- IP Telephony System 10
- A Typical Scenario 11
- The Back Panel 13
- The Front Panel 14
- Chapter 4: Getting Started 15
- Internal Calls 16
- Connect the Internet Phones 18
- External Calls 19
- Action Command 22
- (press these 22
- Choices Description 22
- Entering a Password 26
- Auto-Attendant Messages 27
- 1 to Record 28
- 2 to Review 28
- 3 to Delete 28
- * to Exit 28
- The PBX Status Screen 30
- Parking Lot 31
- Line 1 Calls 31
- The Router Tab 32
- The Router - WAN Setup Screen 34
- MAC Clone Settings 35
- Remote Management 35
- QOS Settings 35
- The Router - LAN Setup Screen 36
- Port Forwarding Settings 37
- DMZ Settings 37
- System Reserved Ports Range 37
- The Voice Tab 38
- Line 1/2/3/4 Status 40
- Auto Attendant Prompt Status 40
- The Voice - System Screen 41
- The Voice - SIP Screen 42
- SIP Timer Values (sec) 43
- Response Status Code Handling 44
- RTP Parameters 44
- SDP Payload Types 44
- NAT Support Parameters 45
- PBX Parameters 46
- Auto Attendant Parameters 49
- PBX Phone Parameters 51
- Configuration Profile 52
- Firmware Upgrade 53
- General Purpose Parameters 54
- The Voice - Regional Screen 55
- Distinctive Ring Patterns 56
- Control Timer Values (sec) 58
- Miscellaneous 63
- The Voice - FXS 1/2 Screen 66
- Subscriber Information 67
- Dial Plan 68
- Streaming Audio Server (SAS) 68
- Call Feature Settings 68
- Audio Configuration 69
- Network Settings 72
- SIP Settings 72
- NAT Settings 75
- Proxy and Registration 75
- Appendix A: Troubleshooting 77
- For Windows 2000: 80
- For Windows XP: 80
- Ending Ports 84
- Protocol IP Address 84
- For Netscape 4.7 or higher: 86
- Frequently Asked Questions 87
- 4 “Goodbye.” 90
- Recording a New Prompt 91
- Advanced Users 94
- AA Dial Plan 1 dp1 96
- AA Dial Plan 2 dp2 96
- Internal Dial Plan 97
- 4“Goodbye.” 98
- Parameter Description Type 101
- Parameter Value 103
- Overview 106
- Before You Begin 106
- Ethernet Adapter 109
- Appendix F: Windows Help 111
- Appendix G: Glossary 112
- Appendix H: Specifications 117
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