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Page 66 / 127 Scroll up to view Page 61 - 65
59
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
excl. NTFY
to log the start-line only for all messages except NOTIFY requests/responses. Select
1-line excl. REG
to log the start-line only for all messages except REGISTER requests/responses. Select
1-line excl.
OPT|NTFY|REG
to log the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER
requests/responses. Select
full
to log all SIP messages in full text. Select
full excl. OPT
to log all SIP messages
in full text except OPTIONS requests/responses. Select
full excl. NTFY
to log all SIP messages in full text except
NOTIFY requests/responses. Select
full excl. REG
to log all SIP messages in full text except REGISTER
requests/responses. Select
full excl. OPT|NTFY|REG
to log all SIP messages in full text except for OPTIONS,
NOTIFY, and REGISTER requests/responses. The default is
none
.
RTP Log Intvl
. Periodically, the System will log RTP statistics via syslog, depending on debug level. Enter the
period of time in seconds. The default is
0
.
Restrict Source IP
. If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled,
then the proxy IP address for Lines 1 and 2 is treated as an acceptable IP address for both lines. To enable the
Restrict Source IP feature, select
yes
. Otherwise, select
no
. The default is
no
.
Referor Bye Delay
. This controls when the System sends BYE to terminate stale call legs upon 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. The default is
4
.
Refer Target Bye Delay
. For the Refer Target Bye Delay, enter the appropriate period of time in seconds. The
default is
0
.
Referee Bye Delay
. For the Referee Bye Delay, enter the appropriate period of time in seconds. The default is
0
.
Refer-To Target Contact
. To contact the refer-to target, select
yes
. Otherwise, select
no
. The default is
no
.
Sticky 183
. If this feature is enabled, then the IP Telephony will ignore further 180 SIP responses after receiving
the first 183 SIP response for an outbound INVITE. To enable this feature, select
yes
. Otherwise, select
no
. The
default is
no
.
Subscriber Information
Display Name
. Enter the display name for caller ID.
User ID
. Enter the extension number for this line.
Figure 6-38: Voice - FXS 1 Screen - Subscriber
Information
Page 67 / 127
60
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
Dial Plan
Dial Plan
. Enter the dial plan script for this line. Refer to “Appendix C: Dial Plan and Auto-Attendant Scripting for
Advanced Users” for more details.
Streaming Audio Server (SAS)
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. The default is
no
.
SAS DLG Refresh Intvl
. If this is not zero, it is the interval at which the streaming audio server sends out session
refresh (SIP re-INVITE) messages to determine if the connection to the caller is still active. If the caller does not
respond to the refresh message, then the System will end this call with a SIP BYE message. The range is 0 to
255 seconds (0 means that the session refresh is disabled).The default is
30
.
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 will be used by the System’s streaming audio server line
in the SDP of its 200 response to an inbound INVITE message from a client.
Call Feature Settings
Blind Attn-Xfer Enable
. This settings lets the System 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 System
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
. The default is
no
.
MOH Server
. Enter the user ID or URL of the auto-answering streaming audio server. When only a user ID is
specified, the current or outbound proxy will be contacted. Music-on-hold is disabled if the MOH Server is not
specified.
Xfer When Hangup Conf
. This setting makes the System perform a transfer when a conference call has ended.
Select
yes
or
no
from the drop-down menu. The default is
yes
.
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, e.g.,
or
conf
(which uses the Proxy value as the domain).
Conference Bridge Ports
. Select the maximum number of conference call participants. The range is 3 to 10. The
default is
3
.
Figure 6-41: Voice - FXS 1 Screen - Call Feature Settings
Figure 6-39: Voice - FXS 1 Screen - Dial Plan
Figure 6-40: Voice - FXS 1 Screen - Streaming Audio
Server
Page 68 / 127
61
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
Enable IP Dialing
. To use IP dialing, select
yes
. Otherwise, select
no
. The default is
no
.
Emergency Number
. This is a comma-separated list of emergency number patterns. If the outbound call
matches one of the patterns, then the System will disable hook flash event handling. Hook flash event handling
will be restored to normal when the phone is on-hook again. If you leave this field blank, then the System will
have no emergency number.
Mailbox ID
. Enter the ID number of the mailbox for this line.
Audio Configuration
Preferred Codec
. Select a 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
. The default is
G711u
.
Silence Supp Enable
. To enable silence suppression so that silent audio frames are not transmitted, select
yes
.
Otherwise, select
no
. The default is
no
.
Use Pref Codec Only
. To only use the preferred codec for all calls, select
yes
. (The call will fail if the far end does
not support this codec.) Otherwise, select
no
. The default is
no
.
Silence Threshold
. Select the appropriate setting for the threshold:
high
,
medium
, or
low
. The default is
medium
.
G729a Enable
. To enable the use of the G729a codec at 8kbps, select
yes
. Otherwise, select
no
. The default
is
yes
.
Echo Canc Enable
. To enable the use of the echo canceller, select
yes
. Otherwise, select
no
. The default is
yes
.
G723 Enable
. To enable the use of the G723a codec at 6.3kbps, select
yes
. Otherwise, select
no
. The default
is
yes
.
Echo Canc Adapt Enable
. To enable the echo canceller to adapt, select
yes
. Otherwise, select
no
. The default
is
yes
.
G726-16 Enable
. To enable the use of the G726 codec at 16kbps, select
yes
. Otherwise, select
no
. The default
is
yes
.
Echo Supp Enable
. To enable the use of the echo suppressor, select
yes
. Otherwise, select
no
. The default
is
yes
.
Figure 6-42: Voice - FXS 1 Screen - Audio Configuration
Page 69 / 127
62
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
G726-24 Enable
. To enable the use of the G726 codec at 24kbps, select
yes
. Otherwise, select
no
. The default
is
yes
.
FAX CED Detect Enable
. To enable detection of the fax Caller-Entered Digits (CED) tone, select
yes
. Otherwise,
select
no
. The default is
yes
.
G726-32 Enable
. To enable the use of the G726 codec at 32kbps, select
yes
. Otherwise, select
no
. The default
is
yes
.
FAX CNG Detect Enable
. To enable detection of the fax Calling Tone (CNG), select
yes
. Otherwise, select
no
. The
default is
yes
.
G726-40 Enable
. To enable the use of the G726 codec at 40kbps, select
yes
. Otherwise, select
no
. The default
is
yes
.
FAX Passthru Codec
. Select the codec for fax passthrough,
G711u
or
G711a
. The default is
G711u
.
DTMF Process INFO
. To use the DTMF process info feature, select
yes
. Otherwise, select
no
. The default is
yes
.
FAX Codec Symmetric
. To force the System to use a symmetric codec during fax passthrough, select
yes
.
Otherwise, select
no
. The default is
yes
.
DTMF Process AVT
. To use the DTMF process AVT feature, select
yes
. Otherwise, select
no
. The default is
yes
.
FAX Passthru Method
. Select the fax passthrough method:
None
,
NSE
, or
ReINVITE
. The default is
NSE
.
DTMF Tx Method
. Select the method to transmit DTMF signals to the far end:
InBand
,
AVT
,
INFO
,
Auto
,
InBand+INFO
, or
AVT+INFO
. InBand sends DTMF using the audio path. AVT sends DTMF as AVT events. INFO
uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation.The default is
Auto
.
FAX Process NSE
. To use the fax process NSE feature, select
yes
. Otherwise, select
no
. The default is
yes
.
Hook Flash Tx Method
. Select the method for signaling hook flash events:
None
,
AVT
, or
INFO
. None does not
signal hook flash events. AVT uses RFC2833 AVT (event = 16). INFO uses SIP INFO with the single line signal=hf in
the message body. The MIME type for this message body is taken from the Hook Flash MIME Type setting. The
default is
None
.
FAX Disable ECAN
. If enabled, this feature will automatically disable the echo canceller when a fax tone is
detected. To use this feature, select
yes
. Otherwise, select
no
. The default is
no
.
Page 70 / 127
63
Chapter 6: Using the Web-based Utility
The Voice Tab
IP Telephony System
Release Unused Codec
. This feature allows the release of codecs not used after codec negotiation on the first
call, so that other codecs can be used for the second line. To use this feature, select
yes
. Otherwise, select
no
.
The default is
yes
.
FAX Enable T38
. To enable the use of the ITU-T T.38 standard for faxing, select
yes
. Otherwise, select
no
. The
default is
yes
.
FXS Port Polarity Configuration
Idle Polarity
. Select the polarity before a call is connected,
Forward
or
Reverse
. The default is
Forward
.
Caller Conn Polarity
. Select the polarity after an outbound call is connected,
Forward
or
Reverse
. The default is
Forward
.
Callee Conn Polarity
. Select the polarity after an inbound call is connected,
Forward
or
Reverse
. The default is
Forward
.
When you have finished making changes, click the
Submit All Changes
button to save the changes, or click the
Undo All Changes
button to undo your changes.
Figure 6-43: Voice - FXS 1 Screen - FXS Port Polarity
Configuration

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