Installing the SIP Paging Amplifier
SIP Paging Amplifier Setup
Operations Guide
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CyberData Corporation
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Note
For specific server configurations, go to the following website address:
https://www.cyberdata.net/pages/connecting-to-ip-pbx-servers
Note
The maximum number of total characters in the dial-out field is 64.
RTP Settings
RTP Port (even)
Specify the port number used for the RTP stream after establishing a SIP call. This
port number must be an even number and defaults to 10500. The supported range is
0-65536. Enter up to 5 digits.
Jitter Buffer
Specify the size of the jitter buffer (in milliseconds) used for SIP calls. Valid values are
50-1000.
Call Disconnection
Terminate Call After Delay
Automatically terminate an active call after a given delay in seconds. A value of 0 will
disable this function. Enter up to 8 digits.
Codec Selection
Force Selected Codec
When configured, this option will allow you to force the device to negotiate for the
selected codec. Otherwise, the device will perform codec negotiation using the default
list of supported codecs.
Codec
Select the desired codec (only one may be chosen).
Click the
Save
button to save your configuration settings.
Note
: You need to reboot for changes to take effect.
Click on the
Reboot
button to reboot the system.
Click on the
Toggle Help
button to see a short description of some of the web page
items. First click on the
Toggle Help
button, and you will see a question mark (
)
appear next to some of the web page items. Move the mouse pointer to hover over a
question mark to see a short description of a specific web page item.
Table 2-17. SIP Page Parameters (continued)
Web Page Item
Description
