VTech ET685 Administrator And Provisioning Manual page 158

Setting
Use SIP Compact
Headers
Listen on SIP TCP port
Register HTTP contact
Disable blind transfer
(REFER)
Disable deflection (code
302)
Show History-Info
Show Diversion
Use NAPTR on SIP URIs
RTCP-XR Report Format
Web User Interface (WebUI) Reference
ET685 Administrator and Provisioning Manual
Description
In order to let the phone generate short compact SIP
headers this option should be enabled. Otherwise the old
usual style of SIP headers will be generated.
By default, the phone doesn't on the network_id_port for
TCP connections. To change this behaviour, enable this
option.
This setting decides if the phone must add the http URL of
the phone as additional contact information. WARNING:
Turning this setting on may cause a complete loss of VoIP
ability if the proxy/registrar does not support it. We urge you
strongly to leave it on "off" if you are not absolutely sure that
it is supported by your proxy/registrar.
A boolean to disable blind transfer. If it is on, instead of blind
transfer, on hitting the transfer key, the only call is put on
hold and a prompt offered to make second call and a normal
consultative transfer would follow. This setting was
introduced for PBXs that dont support REFER.
A boolean to stop 3xx codes (e.g. 302 Moved temporarily).
If the setting is on, a Busy Here is returned. Turning this
setting on will also disable Call Deflect.
When this feature is set to "on", the phone shows the
information available through History-Info header in the
incoming INVITE.
When this feature is set to "on", the phone shows the
information available through Diversion header in the
incoming INVITE.
When this feature is set to "on", the phone converts SIP uri's
according to the regular expression dialplan of the active
outgoing line for numbers dialed through Received and
Missed call lists. For normal phone operation it is best to
leave it turned off, as a valid SIP uri need not be converted
again. Only valid if the pbx used can not append the
requisite leading digits to reach remote destination or if the
number does not already contain the extra digits needed.
e.g. adding 00 for an international call or 0 to access a
number outside the local network.
Specifies which parts the voice quality report should be
composed of. The report is encapsulated in a SIP PUBLISH
message that is send if a call is terminated. See also
parameter vq_report_collector.
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