Version 6.2 393 February 2011
SIP User's Manual 10. Configuration Parameters Reference
Parameter Description
header. If P-Asserted-Identity is selected, the Privacy header is
checked and if the Privacy is set to 'id', the calling number is
assumed restricted.
'FROM' = Use the source number received in the From header.
[SelectSourceHeaderFor
CalledNumber]
Determines the SIP header used for obtaining the called number
(destination) for IP-to-Tel calls.
[0] Request-URI header (default) = Obtains the destination number
from the user part of the Request-URI.
[1] To header = Obtains the destination number from the user part of
the To header.
[2] P-Called-Party-ID header = Obtains the destination number from
the P-Called-Party-ID header.
Web/EMS: Forking
Handling Mode
[ForkingHandlingMode]
Determines how the device handles the receipt of multiple SIP 18x
forking responses, for Tel-to-IP calls. The forking 18x response is the
response with a different to-tag than the previous 18x response. Those
responses usually are generated by Proxy/Application servers that
perform call forking, sending the device's originating INVITE to several
destinations, using the same CallID.
[0] Parallel handling = If SIP 18x with SDP is received, the device
opens a voice stream according to the received SDP and disregards
any 18x forking responses (with or without SDP) received thereafter.
If the first response is 180 without SDP, the device responds
according to the PlayRBTone2TEL parameter and disregards the
subsequent forking 18x responses. (default)
[1] Sequential handling = If 18x with SDP is received, the device
opens a voice stream according to the received SDP. The device re-
opens the stream according to subsequently received 18x responses
with SDP, or plays a ringback tone if 180 response without SDP is
received. If the first received response is 180 without SDP, the
device responds according to the PlayRBTone2TEL parameter and
processes the subsequent 18x forking responses.
Note: Regardless of this parameter setting, once a SIP 200 OK
response is received, the device uses the RTP information and re-
opens the voice stream, if necessary.
Web: Forking Timeout
[ForkingTimeOut]
The timeout (in seconds) that is started after the first SIP 2xx response
has been received for a User Agent when a Proxy server performs call
forking (Proxy server forwards the INVITE to multiple SIP User Agents).
The device sends a SIP ACK and BYE in response to any additional
SIP 2xx received from the Proxy within this timeout. Once this timeout
elapses, the device ignores any subsequent SIP 2xx.
The number of supported forking calls per channel is 4. In other words,
for an INVITE message, the device can receive up to 4 forking
responses from the Proxy server.
The valid range is 0 to 30. The default is 30.
Web/EMS: Enable Reason
Header
[EnableReasonHeader]
Enables or disables the usage of the SIP Reason header.
[0] Disable
[1] Enable (default)
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