Matrix PRASAR UCS System Manual 271
When extension's state is changed from Ringing (early state as defined in BLF) to Mature (confirm state,
as defined in BLF) state, because of implementation of PRASAR UCS, it will send 'Terminate' state while
moving from ringing to mature state. The interpretation of terminate message will vary from terminal to
terminal.
• To allow the SIP Extension to view the status of other SIP-enabled Terminals, whether they are available or
not, select the Presence Subscription check box. Default: disabled.
The SIP Extension, for which you have enabled Presence Subscription, will be able to view Presence of
only those SIP Extensions which have PUBLISH enabled.
• To allow the SIP Extension to publish presence using the PUBLISH feature supported by the SIP
Extension, select the PUBLISH Enable check box. Default: disabled.
By default, Authentication for PUBLISH message is enabled. You may disable if you do not want to use
Authentication.
You must configure the Authentication ID, if you have enabled both Publish and Authentication.
• For secure conversations over SIP, enable SRTP Mode. The PRASAR UCS supports the following
options:
• Disable: PRASAR UCS uses normal RTP for transporting the speech packets.
• Optional: PRASAR UCS uses SRTP for transporting the speech packets. If the remote user does not
support SRTP, PRASAR UCS will use normal RTP for transporting the speech packets.
• If you select this option, you must configure the SRTP Media Type. You may select AVP or SAVP.
By default, AVP is selected as the SRTP Media Type.
• Forced: PRASAR UCS uses only SRTP (SAVP) for transporting the speech packets. If the remote user
does not support SRTP, PRASAR UCS will reject incoming calls from and drop outgoing calls made to
such users.
By default, SRTP Mode is Disabled.
• Key Templates are not applicable to Standard SIP Phones registered with PRASAR UCS.
• Assign a SIP Hardware Template to the SIP Extension. Default: 01. The “SIP Hardware Template”
contains voice quality related features such as Voice Codec selection, Tx and Rx Gains, Echo
Cancellation, Jitter Buffer and Fax-over-IP options and related parameters
There are 32 different templates to choose from. Each template can also be altered to suit your
requirement and preferences. By default, Template number 01 is assigned to all SIP Extensions as well as
to SIP Trunks.
Check if the values in this template fulfill requirements of the SIP Extension. If Template 01 fulfills the
feature requirements, retain Template 01.
If a different set of SIP hardware features are to be allowed to this SIP Extensions, prepare another
template and assign it to this extension. To do this,
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