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These settings are shown for SIP IP extensions.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field | Description | ||||
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IP Address |
Default = 0.0.0.0
The IP address of the phone. The default setting accepts connection from any address. If an address is entered, registration is only accepted from a device with that address. |
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Codec Selection |
Default = System Default
This field defines the codec or codecs offered during call setup. The available codecs in default preference order are: G.711 A-Law, G.711 ULAW, G.729 and G.723.1. Note that the default order for G.711 codecs will vary to match the system's default companding setting. G.723.1 is not supported on Linux based systems. The G.722 64K codec is also supported on IP500/IP500 V2 systems with IP500 VCM, IP500 VCM V2 or IP500 Combo cards. For Server Edition it is supported on Primary Server, Secondary Serverand Expansion System (L) systems and on Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo. The codecs available to be used are set through the System Codec list (System | System Codec). The options are:
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Fax Transport Support: |
Default = Off.
This option is only available if Re-Invite Supported is selected. When enabled, the system performs fax tone detection on calls routed via the line and, if fax tone is detected, renegotiates the call codec as configured below. The SIP line provider must support the selected fax method and Re-Invite. The system must have available VCM resources using an IP500 VCM, IP500 VCM V2 or IP500 Combo base card. For systems in a network, fax relay is supported for fax calls between the systems. The options are:
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TDM | IP Gain |
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP connection. This field is not shown on Linux based platforms. |
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IP | TDM Gain | Default = Default (0dB). Range = -31dB to +31dB. Allows adjustment of the gain on audio from the IP connection to the system TDM interface. This field is not shown on Linux based platforms. | ||||
DTMF Support |
Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled to the remote end. The supported options are In Band, RFC2833 or Info. |
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3rd Party Auto Answer |
Default = None.
This setting applies to 3rd party standard SIP extensions. The options are:
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Media Security |
Default = Same as System.
These settings control whether SRTP is used for this extension and the settings used for the SRTP. The options are:
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Advanced Media Security Options |
Not displayed if Media Security is set to Disabled. The
options are:
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VoIP Silence Suppression |
Default = Off
When selected, this option will detect periods of silence on any call over the line and will not send any data during those silent periods. This feature is not used on IP lines using G.711 between systems. On trunk's between networked systems, the same setting should be set at both ends |
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Local Hold Music | Default = Off. | ||||
Allow Direct Media Path |
Default = On.
This settings controls whether IP calls must be routed via the system or can be routed alternately if possible within the network structure If enabled, IP calls can take routes other than through the system. This removes the need for a voice compression channel. Both ends of the calls must support Direct Media and be using the same protocol (H.323 or SIP). Enabling this option may cause some vendors problems with changing the media path mid call. If disabled or not supported at on one end of the call, the call is routed via the system. RTP relay support allows calls between devices using the same audio codec to not require a voice compression channel. |
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RE-Invite Supported |
Default = On.
When enabled, Re-Invite can be used during a session to change the characteristics of the session. For example when the target of an incoming call or a transfer does not support the codec originally negotiated on the trunk. Requires the ITSP to also support Re-Invite. |
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Codec Lockdown |
Default = Off.
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In response to a SIP offer with a list of codecs supported, some SIP user agents supply a SDP answer that also lists multiple codecs. This means that the user agent may switch to any of those codecs during the session without further negotiation. The system does not support multiple concurrent codecs for a session, so loss of speech path will occur if the codec is changed during the session. If codec lockdown is enabled, when the system receives an SDP answer with more than one codec from the list of offered codecs, it sends an extra re-INVITE using just a single codec from the list and resubmits a new SDP offer with just the single chosen codec. |
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Reserve License |
Default = None. Each Avaya IP phones requires an Avaya IP Endpoint
license. Each non-Avaya IP phones requires an 3rd Party IP Endpoint
license. Normally these licenses are issued in the order that devices
register. This option allows this extension to be pre-licensed before
the device has registered. This helps prevent a previously licensed
phone becoming unlicensed following a system restart if unlicensed
devices are also present. The options are:
Note the following:
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