SIP User's Manual 608 Document #: LTRT-83310
Mediant 600 & Mediant 1000
Parameter Description
The "From" and "Pai2" values are not case-sensitive.
Once a URL is selected, all the calling party parameters are set
from this header. If P-Asserted-Identity is selected and the Privacy
header is set to 'id', the calling number is assumed restricted.
[SelectSourceHeaderForCa
lledNumber]
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 SIP to-tag than the previous 18x response.
These responses are typically generated (initiated) by Proxy /
Application servers that perform call forking, sending the device's
originating INVITE (received from SIP clients) 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]
Defines 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 20. In other
words, for an INVITE message, the device can receive up to 20
forking responses from the Proxy server.
The valid range is 0 to 30. The default is 30.
Web: Tel2IP Call Forking
Mode
[Tel2IPCallForkingMode]
Enables Tel-to-IP call forking, whereby a Tel call can be routed to
multiple IP destinations.
[0] Disable (default)
[1] Enable
Note: Once enabled, routing rules must be assigned Forking Groups
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