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SPA9000 Field Reference
Voice tab
SPA9000 Voice System Administration Guide
244
B
Voice tab > FXS 1/2 page
Subscriber Information section
Voice tab > FXS 1/2 page
Dial Plan section
Referor Bye Delay
Controls when the SPA9000 sends BYE to terminate stale call
legs after completion of call transfers. Multiple delay settings
(Referor, Refer Target, Referee, and Refer-To Target) are
configured on this screen. For the Referor Bye Delay, enter
the appropriate period of time in seconds.
Default:
4
Refer Target Bye Delay
For the Refer Target Bye Delay, enter the appropriate period
of time in seconds.
Default:
0
Referee Bye Delay
For the Referee Bye Delay, enter the appropriate period of
time in seconds.
Default:
0
Refer-To Target Contact
To contact the refer-to target, select yes. Otherwise, select
no.
Default:
no
Sticky 183
If this feature is enabled, the IP telephony ignores further 180
SIP responses after receiving the first 183 SIP response for
an outbound INVITE. To enable this feature, select yes.
Otherwise, select no.
Default:
no
Display Name
Display name for caller ID.
User ID
Extension number for this line.
Dial Plan
Dial plan script for this line.
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SPA9000 Field Reference
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SPA9000 Voice System Administration Guide
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Voice tab > FXS 1/2 page
Mailbox Status section
Voice tab > FXS 1/2 page
Streaming Audio Server (SAS) section
Message Waiting
Indicates whether you have new voicemail waiting.
SAS Enable
To enable the use of the line as a streaming audio source,
select yes. Otherwise, select no. If enabled, the line cannot
be used for outgoing calls. Instead, it auto-answers incoming
calls and streams audio RTP packets to the caller.
Default:
no
SAS DLG Refresh Intvl
If this value is not zero, it is the interval at which the streaming
audio server sends out session refresh (SIP re-INVITE)
messages to determine whether the connection to the caller
is still active. If the caller does not respond to the refresh
message, the SPA9000 ends this call with a SIP BYE
message. The range is 0 to 255 seconds (0 means that the
session refresh is disabled).
Default:
30
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SPA9000 Voice System Administration Guide
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Voice tab > FXS 1/2 page
Call Feature Settings section
SAS Inbound RTP Sink
This setting works around devices that do not play inbound
RTP if the streaming audio server line declares itself as a
send-only device and tells the client not to stream out audio.
Enter a Fully Qualified Domain Name (FQDN) or IP address of
an RTP sink; this value is used by the SPA9000’s streaming
audio server line in the SDP of its 200 response to an inbound
INVITE message from a client.
The purpose of this parameter is to work around devices that
do not play inbound RTP if the SAS line declares itself as a
“sendonly” device and tells the client not to stream out audio.
This parameter is a FQDN or IP address of a RTP sink to be
used by the SPA SAS line in the SDP of its 200 response to
inbound INVITE from a client. It will appear in the c = line and
the port number and, if specified, in the m = line of the SDP.
If
this value is not specified or equal to 0, then c = 0.0.0.0 and
a=sendonly is used in the SDP to tell the SAS client to not to
send any RTP to this SAS line. If a non-zero value is specified,
then a=sendrecv and the SAS client will stream audio to the
given address. Special case: If the value is $IP, then the SAS
line’s own IP address is used in the c = line and a=sendrecv. In
that case the SAS client will stream RTP packets to the SAS
line.
Default: blank
Blind Attn-Xfer Enable
Enables the SPA9000 to perform an attended transfer
operation by ending the current call leg and performing a
blind transfer of the other call leg. If this feature is disabled,
the SPA9000 performs an attended transfer operation by
referring the other call leg to the current call leg while
maintaining both call legs. To use this feature, select yes.
Otherwise, select no.
Default:
no
MOH Server
User ID or URL of the auto-answering streaming audio server.
When only a user ID is specified, the current or outbound
proxy is contacted. Music-on-hold is disabled if the MOH
Server is not specified.
Xfer When Hangup Conf
Makes the SPA9000 perform a transfer when a conference
call has ended. Select yes or no from the drop-down menu.
Default:
Yes
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SPA9000 Field Reference
Voice tab
SPA9000 Voice System Administration Guide
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Voice tab > FXS 1/2 page
Audio Configuration section
NOTE
A codec resource is considered as allocated if it has been included in the SDP
codec list of an active call, even though it eventually may not be the one chosen for
the connection. So, if the G.729a codec is enabled and included in the codec list,
that resource is tied up until the end of the call whether or not the call actually uses
G.729a. If the G729a resource is already allocated and since only one G.729a
resource is allowed per device, no other low-bit-rate codec may be allocated for
subsequent calls; the only choices are G711a and G711u. On the other hand, two
G.723.1/G.726 resources are available per device.Therefore it is important to
disable the use of G.729a in order to guarantee the support of two simultaneous
G.723/G.726 codecs.
Conference Bridge URL
This feature supports external conference bridging for n-way
conference calls (n > 2), instead of mixing audio locally. To use
this feature, set this parameter to that of the server's name;
for example,
[email protected]:12345 or conf
(which uses the Proxy value as the domain).
Default: blank
Conference Bridge Ports
Maximum number of conference call participants. The range
is 3 to 10.
Default:
3
Enable IP Dialing
To use IP dialing, select yes. Otherwise, select no.
Default:
no
Emergency Number
Comma-separated list of emergency number patterns. If an
outbound call matches one of the patterns, the SPA9000
disables call waiting, hook flash event handling, and fax tone
detection. Operation is restored to normal when the phone is
on-hook again. If you leave this field blank, the SPA9000 has
no emergency number.
Mailbox ID
Enter the ID number of the mailbox for this line.
Preferred Codec
Preferred codec for all calls. (The actual codec used in a call
still depends on the outcome of the codec negotiation
protocol.) Select one of the following:
G711u
,
G711a
,
G726-
16
,
G726-24
,
G726-32
,
G726-40
,
G729a
, or
G723
Default:
G711u
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SPA9000 Field Reference
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SPA9000 Voice System Administration Guide
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Second Preferred Codec,
Third Preferred Codec
Other codecs to be preferred if the preferred codec is not
available due to the codec negotiation protocol.
Default:
Unspecified
Use Pref Codec Only
If set to yes, the call can be completed only with the
preferred codec.
Default:
no
Silence Supp Enable
To enable silence suppression so that silent audio frames are
not transmitted, select yes. Otherwise, select no.
Default:
no
Use Pref Codec Only
To use only the preferred codec for all calls, select yes. (The
call fails if the far end does not support this codec.)
Otherwise, select no.
Default:
no
Silence Threshold
Select the appropriate setting for the threshold: high,
medium, or low.
Default:
medium
G729a Enable
To enable the use of the G729a codec at 8 kbps, select yes.
Otherwise, select no.
Default:
yes
Echo Canc Enable
To enable the use of the echo canceller, select yes.
Otherwise, select no.
Default:
yes
G723 Enable
To enable the use of the G723a codec at 6.3 kbps, select yes.
Otherwise, select no.
Default:
yes
Echo Canc Adapt Enable
To enable the echo canceller to adapt, select yes. Otherwise,
select no.
Default:
yes
G726-16 Enable
To enable the use of the G726 codec at 16 kbps, select yes.
Otherwise, select no.
Default:
yes
Echo Supp Enable
To enable the use of the echo suppressor, select yes.
Otherwise, select no.
Default:
yes
G726-24 Enable
To enable the use of the G726 codec at 24 kbps, select yes.
Otherwise, select no.
Default:
yes
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