Use the Voice > Information page to view information about the ATA voice application.
To open this page: Click Voice on the menu bar, and then click Information in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
Product Information
Field
Description
Product Name
Model number/name.
Serial Number
Product serial number.
Software Version
Software version number.
Hardware Version
Hardware version number.
MAC Address
MAC Address. For example: 8843E1657936.
Client Certificate
Status of the client certificate, which can indicate if the ATA was authorized by your ITSP.
Customization
Used for Remote Configuration by service providers who deploy the ATA to their customers.
•Open: Not a Remote Configuration unit. This ATA can be configured by using the configuration utility.
•Pending: This Remote Configuration unit has not yet connected to the server to get its profile.
•Customized: This Remote Configuration unit has received its profile from the server.
System Status
Field
Description
Current Time
Current date and time of the system; for example, 10/3/2003 16:43:00. Set the system time by using the Network Setup > Time Settings page.
Elapsed Time
Total time elapsed since the last reboot of the system; for example, 25 days and 18:12:36.
RTP Packets Sent
Total number of RTP packets sent (including redundant packets)
RTP Bytes Sent
Total number of RTP bytes sent.
RTP Packets Recv
Total number of RTP packets received (including redundant packets)
RTP Bytes Recv
Total number of RTP bytes received.
SIP Messages Sent
Total number of SIP messages sent (including retransmissions)
SIP Bytes Sent
Total number of bytes of SIP messages sent (including retransmissions)
SIP Messages Recv
Total number of SIP messages received (including retransmissions)
SIP Bytes Recv
Total number of bytes of SIP messages received (including retransmissions)
External IP
The External IP address used for NAT mapping.
Line 1 Status
Field
Description
Hook State
The hook state of the port: On or Off.
Registration State
Indicates if the line has registered with the SIP proxy.
Last Registration At
Last date and time the line was registered.
Next Registration In
The number of seconds before the next registration renewal. Indicates whether you have new voice mail waiting.
Message Waiting
Indicates Yes when a message is received.
Mapped SIP Port
Port number of the SIP port mapped by NAT.
Call Back Active
Indicates whether or not a call back request is in progress. Options are either yes or no.
Last Called Number
The phone number that was most recently called through this port.
Last Caller Number
The originating phone number of the call that was most recently received through this port.
Call 1 and 2 State
Indicates the state of calls, if any:
•Idle
•Collecting PSTN PIN
•Invalid PSTN PIN
•PSTN Caller Accepted
•Connected to PSTN
Call 1 and 2 Tone
The type of tone used by the call.
Call 1 and 2 Encoder
The codec used for encoding.
Call 1 and 2 Decoder
The codec used for decoding.
Call 1 and 2 FAX
The status of the fax passthrough mode.
Call 1 and 2 Type
The direction of the call. May take one of the following values:
•PSTN Gateway Call = VoIP-To-PSTN Call
•VoIP Gateway Call = PSTN-To-VoIP Call
•PSTN To Line 1 = PSTN call ring through and answered by Line 1
•Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW
•Line 1 Forward to PSTN Number =VoIP calls Line 1 then forwarded to PSTN number
•Line 1 To PSTN Gateway
•Line 1 Fallback To PSTN Gateway
Call 1 and 2 Remote Hold
Indicates whether the far end has placed the call on hold.
Call 1 and 2 Callback
Indicates whether the call was triggered by a call back request.
Call 1 and 2 Peer Name
The name of the peer phone.
Call 1 and 2 Peer Phone
The phone number of the peer phone.
Call 1 and 2 Call Duration
The duration of the call.
Call 1 and 2 Packets Sent
The number of packets sent
Call 1 and 2 Packets Recv
The number of packets received.
Call 1 and 2 Bytes Sent
The number of bytes sent.
Call 1 and 2 Bytes Recv
The number of bytes received.
Call 1 and 2 Decode Latency
The number of milliseconds for decoder latency.
Call 1 and 2 Jitter
The number of milliseconds for receiver jitter
Call 1 and 2 Round Trip Delay
The number of milliseconds for delay.
Call 1 and 2 Packets Lost
The number of packets lost.
Call 1 and 2 Packet Error
The number of invalid packets received.
Custom CA Status
Field
Description
Custom CA Provisioning Status
The status of the latest custom CA (Certificate Authority) certificate download.
Custom CA Info
The successfully downloaded CA information, or "Not Installed" if no custom CA certificate was installed. Default setting: Not Installed
PSTN Line Status
Field
Description
Hook State
The hook state of the LINE port: On or Off.
Line Voltage
The voltage existing on the PSTN line.
Loop Current
The current (milliamperes) existing on the local loop.
Registration State
Indicates if the line has registered with the SIP proxy.
Last Registration At
The last date and time when the line was registered.
Next Registration In
The number of seconds before the next registration renewal.
Last Called VoIP Number
The VoIP phone number that was most recently called through this port.
Last Called PSTN Number
The PSTN phone number that was most recently called through the LINE port
Last VoIP Caller
The originating phone number of the VoIP call that was most recently received through the LINE port.
Last PSTN Caller
The originating phone number of the PSTN call that was most recently received through the LINE port.
Last PSTN Disconnect Reason
The reason for the ATA hanging up the LINE port:
•PSTN Disconnect Tone
•PSTN Activity Timeout
•CPC Signal
•Polarity Reversal
•VoIP Call Failed
•VoIP Call Ended
•Invalid VoIP Destination
•Invalid PIN
•PIN Digit Timeout
•VoIP Dialing Timeout
•PSTN Gateway Call Timeout
•VoIP Gateway Call Timeout
PSTN Activity Timer
The time in milliseconds (ms) before the ATA disconnects the current gateway unless the PSTN side has some audio activity.
Mapped SIP Port
The port number of the SIP port mapped by NAT.
Call Type
The type of call:
•PSTN Gateway Call = VoIP-To-PSTN Call
•VoIP Gateway Call = PSTN-To-VoIP Call
•PSTN To Line 1 = PSTN call ring through and answered by Line 1
•Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW
•Line 1 Forward to PSTN Number =VoIP calls Line 1 then forwarded to PSTN number
•Line 1 To PSTN Gateway
•Line 1 Fallback To PSTN Gateway
VoIP State
May take one of the following values:
Idle
•Collecting PSTN Pin
•Invalid PSTN PIN
•PSTN Caller Accepted
•Connected to PSTN
PSTN State
May take one of the following values:
Idle
•Collecting PSTN Pin
•Invalid PSTN PIN
•PSTN Caller Accepted
•Connected to PSTN
VoIP Tone
The tone that is being played to the VoIP call leg.
PSTN Tone
The tone that is being played to the PSTN call leg.
VoIP Peer Name
The of the party at the VoIP call leg.
PSTN Peer Name
The name of the party at the PSTN call leg.
VoIP Peer Number
The phone number of the party at the VoIP call leg.
PSTN Peer Number
The phone number of the party at the PSTN call leg.
VoIP Call Encoder
The audio encoder being used for the VoIP call leg.
VoIP Call Decoder
The audio decoder being used for the VoIP call leg.
VoIP Call FAX
The status of the fax passthrough mode for VoIP calls.
VoIP Call Remote Hold
Indicates whether the far end has placed the VoIP call on hold.
VoIP Call Duration
The duration of the VoIP call.
VoIP Call Packets Sent
The number of packets sent for VoIP calls.
VoIP Call Packets Recv
The number of packets received for VoIP calls.
VoIP Call Bytes Sent
The number of bytes sent for VoIP calls.
VoIP Call Bytes Recv
The number of bytes received for VoIP calls.
VoIP Call Decode Latency
The number of milliseconds for decoder latency for VoIP calls.
VoIP Call Jitter
The number of milliseconds for receiver jitter for VoIP calls.
VoIP Call Round Trip Delay
The number of milliseconds for delay for VoIP calls.
VoIP Call Packets Lost
The number of packets lost for VoIP calls.
VoIP Call Packet Error
The number of invalid packets received for VoIP calls.
VoIP Call Mapped RTP Port
The port mapped for Real Time Protocol traffic for VoIP calls.
DECT 1 ~ DECT 10 Status
Field
Description
Registration State
Indicates whether or not the line has registered with the SIP proxy: Registered, Not Registered, or Failed.
Last Registration At
The last date and time when the line was registered.
Next Registration In
The number of seconds before the next registration renewal.
Message Waiting
Indicates whether or not there are new messages: yes or no. The value automatically is set to yes when a message is received. You also can clear or set the flag manually from the User 1 page.
Call Back Active
Indicates whether a call back request is in progress: yes or no.
Last Called Number
The phone number that was most recently called through this port.
Last Caller Number
The originating phone number of the call that was most recently received through this port.
Mapped SIP Port
Port number of the SIP port mapped by NAT.
Call 1 and 2 State
The current call state:
•Idle
•Collecting PSTN Pin
•Invalid PSTN PIN
•PSTN Caller Accepted
•Connected to PSTN
Call 1 and 2 Tone
The type of tone used by the call.
Call 1 and 2 Encoder
The codec used for encoding.
Call 1 and 2 Decoder
The codec used for decoding.
Call 1 and 2 FAX
The status of the fax passthrough mode.
Call 1 and 2 Type
The direction of the call:
•PSTN Gateway Call = VoIP-To-PSTN Call
•VoIP Gateway Call = PSTN-To-VoIP Call
•PSTN To Line 1 = PSTN call ring through and answered by Line 1
•Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW
•Line 1 Forward to PSTN Number =VoIP calls Line 1 then forwarded to PSTN number
•Line 1 To PSTN Gateway
•Line 1 Fallback To PSTN Gateway
Call 1 and 2 Remote Hold
Indicates whether the far end has placed the call on hold.
Call 1 and 2 Callback
Indicates whether the call was triggered by a call back request.
Call 1 and 2 Peer Name
The name of the peer phone.
Call 1 and 2 Peer Phone
The phone number of the peer phone.
Call 1 and 2 Call Duration
The duration of the call.
Call 1 and 2 Packets Sent
The number of packets sent.
Call 1 and 2 Packets Recv
The number of packets received.
Call 1 and 2 Bytes Sent
The number of bytes sent.
Call 1 and 2 Bytes Recv
The number of bytes received.
Call 1 and 2 Decode Latency
The number of milliseconds for decoder latency.
Call 1 and 2 Jitter
The number of milliseconds for receiver jitter
Call 1 and 2 Round Trip Delay
The number of milliseconds for delay.
Call 1 and 2 Packets Lost
The number of packets lost.
Call 1 and 2 Packet Error
The number of invalid packets received.
Call 1 and 2 Mapped RTP Port
The port mapped for Real Time Protocol traffic for Call 1/2.
DECT Handset 1 ~ DECT Handset 10 Status
Field
Description
Handset IPEI
The unique hardware identifier of the unit, comparable to a MAC address.
Model Number
The Cisco model number of the unit.
System
Use the Voice > System page to configure general voice system settings and to enable logging by using a syslog server. (Logging also can be configured in the Administration > Logging pages. For more information, see Logging.)
To open this page: Click Voice on the menu bar, and then click System in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
Requirements for Logging
•You need a computer that is on the same subnetwork as the ATA, to capture the log files. This computer needs to be running a syslog daemon. Enter the IP address of this computer in the Syslog Server and Debug Server fields.
•You can deploy a syslog server to receive syslog messages from the ATA, which acts as a syslog client. The syslog client device uses the syslog protocol to send messages, based on its configuration, to a syslog server. The syslog messages can be accessed by reviewing the "syslog.514.log" file which resides in the same directory as the slogsrv.exe syslog server application.
Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
System Configuration
Field
Description
Restricted Access Domains
Domain of the service provider to which the ATA is connected to. It prevents the ATA from connecting to other service providers.
IVR Admin Password
Password for the administrator to manage the ATA by using the built-in IVR through a connected phone.
NOTE The registered HS can also manage the ATA or deregister other HS with this password.
Network Startup Delay
The number of seconds of delay between restarting the voice module and initializing network interface. Default setting: 3
Handset (HS) Pairing Password
Password used as self authentication for Handset registration and deregistration.
Default setting: blank
NOTE There are 3 options in Handset registration:
· Register: For an unregistered HS, select Register and enter the Handset Pairing Passwd for authentication.
· Deregister: For a registered HS, select Deregister and enter the Handset Pairing Passwd for authentication. Use select softkey to deregister the HS.
· Deregister Others: For a registered HS, select Deregister Others. Enter <IVR Admin Passwd> for authentication. Use select softkey to deregister others.
Miscellaneous Settings
Field
Description
DNS Query TTL Ignore
In DNS packages, the server will suggest a TTL value to the client; if this parameter is set to yes, the value from the server will be ignored. Default setting: yes
Syslog Server
Specify the syslog server name and port. This feature specifies the server for logging ATA system information and critical events. If both Debug Server and Syslog Server are specified, Syslog messages are also logged to the Debug Server. Default setting: blank
Syslog Server Transport
The Syslog messages can be printed to the server with the selected tranport method (UDP or TLS)
Default setting: UDP
Debug Server
The debug server name and port. This feature specifies the server for logging debug information. The level of detailed output depends on the debug level parameter setting. Default setting: blank
Debug Server Transport
The Debug information can be printed to the server with the selected transport method (UDP and TLS.)
Default setting: UDP
Debug Level
Determines the level of debug information that will be generated. Select 0, 1, 2, 3 or 3+Router from the drop-down list. The higher the debug level, the more debug information will be generated. Level 0 means that no information will be collected. Levels 1, 2 & 3 generate messages related to the voice ports only. Level 3+Router generates debug content for both voice and router components. Default setting: 3
DevTest Password
Used for internal automation test; not for the user.
Syslog Prefix
Allows the user to prefix additional information to syslog.
Default setting: 215
DECT Codec Change
Codec change between handset (HS) and base to match RTP codec.
Default setting: yes
SIP
Use the Voice > SIP page to configure SIP parameters and values.
To open this page: Click Voice on the menu bar, and then click SIP in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
NOTE For a deeper understanding of these fields, refer to Request for Comments (RFC) 3261.
SIP Parameters
Field
Description
Max Forward
The maximum times a call can be forwarded. The valid range is from 1 to 255. Default setting: 70
Max Redirection
Number of times an invite can be redirected to avoid an infinite loop. Default setting: 5.
Max Auth
The maximum number of times (from 0 to 255) a request may be challenged. Default setting: 2
SIP User Agent Name
The User-Agent header used in outbound requests. If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed. Default setting: $VERSION
SIP Server Name
The server header used in responses to inbound responses. Default setting: $VERSION
SIP Reg User Agent Name
The User-Agent name to be used in a REGISTER request. If this value is not specified, the SIP User Agent Name parameter is also used for the REGISTER request. Default setting: blank
SIP Accept Language
Accept-Language header used. There is no default (this indicates that the ATA does not include this header) If empty, the header is not included. Default setting: blank
DTMF Relay MIME Type
The MIME Type used in a SIP INFO message to signal a DTMF event. Default setting: application/dtmf-relay.
Hook Flash MIME Type
The MIME Type used in a SIP INFO message to signal a hook flash event. Default setting: application/hook-flash
Remove Last Reg
Determines whether or not the ATA removes the last registration before submitting a new one, if the value is different. Select yes to remove the last registration, or select no to omit this step. Default setting: no
Use Compact Header
Determines whether or not the ATA uses compact SIP headers in outbound SIP messages. Select yes or no from the drop-down list. Select yes to use compact SIP headers in outbound SIP messages. Select no to use normal SIP headers. If inbound SIP requests contain compact headers, the ATA reuses the same compact headers when generating the response regardless the settings of the Use Compact Header parameter. If inbound SIP requests contain normal headers, the ATA substitutes those headers with compact headers (if defined by RFC 261) if Use Compact Header parameter is set to yes. Default setting: no
Escape Display Name
Determines whether or not the Display Name is private. Select yes if you want the ATA to enclose the string (configured in the Display Name) in a pair of double quotes for outbound SIP messages.If the display name includes " or \, these will be escaped to \" and \\ within the double quotes. Otherwise, select no. Default setting: no
RFC 2543 Call Hold
Configures the type of call hold: a:sendonly or 0.0.0.0. Do not use the 0.0.0.0 syntax in a HOLD SDP; use the a:sendonly syntax. Default setting: no
Mark all AVT Packets
Select yes if you want all AVT tone packets (encoded for redundancy) to have the marker bit set for each DTMF event. Select no to have the marker bit set only for the first packet. Default setting: yes
SIP TCP Port Min
The lowest TCP port number that can be used for SIP sessions. Default setting: 5060
SIP TCP Port Max
The highest TCP port number that can be used for SIP sessions. Default setting: 5080
CTI Enable
Enables or disables the Computer Telephone Interface feature provided by some servers. Default setting: no
SIP Timer Values
Field
Description
SIP T1
RFC 3261 T1 value (round-trip time estimate), which can range from 0 to 64 seconds. Default setting: 0.5
SIP T2
RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses), which can range from 0 to 64 seconds. Default setting: 4
SIP T4
RFC 3261 T4 value (maximum duration a message remains in the network), which can range from 0 to 64 seconds. Default setting: 5
SIP Timer B
INVITE time-out value, which can range from 0 to 64 seconds. Default setting: 32
SIP Timer F
Non-INVITE time-out value, which can range from 0 to 64 seconds. Default setting: 32
SIP Timer H
H INVITE final response, time-out value, which can range from 0 to 64 seconds. Default setting: 32
SIP Timer D
ACK hang-around time, which can range from 0 to 64 seconds. Default setting: 32
SIP Timer J
Non-INVITE response hang-around time, which can range from 0 to 64 seconds. Default setting: 32
INVITE Expires
INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Range: 0-(231-1) Default setting: 240
ReINVITE Expires
ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Range: 0-(231-1) Default setting: 30
Reg Min Expires
Minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter. If the proxy returns a value less than this setting, the minimum value is used. Default setting: 1
Reg Max Expires
Maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is larger than this setting, the maximum value is used. Default setting: 7200
Reg Retry Intvl
Interval to wait before the ATA retries registration after failing during the last registration. Default setting: 30
Reg Retry Long Intvl
When registration fails with a SIP response code that does not match Retry Reg RSC, the ATA waits for the specified length of time before retrying. If this interval is 0, the ATA stops trying. This value should be much larger than the Reg Retry Intvl value, which should not be 0. Default setting: 1200
Reg Retry Random Delay
Random delay range (in seconds) to add to Register Retry Intvl when retrying REGISTER after a failure. Default setting: 0 (disabled)
Reg Retry Long Random Delay
Random delay range (in seconds) to add to Register Retry Long Intvl when retrying REGISTER after a failure. Default setting: 0 (disabled)
Reg Retry Intvl Cap
The maximum value to cap the exponential back-off retry delay (which starts at Register Retry Intvl and doubles on every REGISTER retry after a failure) In other words, the retry interval is always at Register Retry Intvl seconds after a failure. If this feature is enabled, Reg Retry Random Delay is added on top of the exponential back-off adjusted delay value. Default setting: 0, which disables the exponential backoff feature.
Response Status Code Handling
Field
Description
SIT1 RSC
SIP response status code for the appropriate Special Information Tone (SIT) For example, if you set the SIT1 RSC to 404, when the user makes a call and a failure code of 404 is returned, the SIT1 tone is played. Reorderor Busytone is played by default for all unsuccessful response status code for SIT 1 RSC through SIT 4 RSC. Default setting: blank
SIT2 RSC
SIP response status code to INVITE on which to play the SIT2 Tone. Default setting: blank
SIT3 RSC
SIP response status code to INVITE on which to play the SIT3 Tone. Default setting: blank
SIT4 RSC
SIP response status code to INVITE on which to play the SIT4 Tone. Default setting: blank
Try Backup RSC
SIP response code that retries a backup server for the current request. Default setting: blank
Retry Reg RSC
Interval to wait before the ATA retries registration after failing during the last registration. Default setting: blank
RTP Parameters
Field
Description
RTP Port Min
Minimum port number for RTP transmission and reception.
The RTP Port Min and RTP Port Max parameters should define a range that contains at least 4 even number ports, such as 100 -106. Default setting: 16384.
RTP Port Max
Maximum port number for RTP transmission and reception. Default setting: 16482.
RTP Packet Size
Packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds. Default setting: 0.030
Max RTP ICMP Err
Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the ATA terminates the call. If value is set to 0, the ATA ignores the limit on ICMP errors. Default setting: 0
RTCP Tx Interval
Interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds. During an active connection, the ATA can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a SR (Sender Report) and a SDES (Source Description) The last RTCP packet contains an additional BYE packet. Each SR except the last one contains exactly 1 RR (Receiver Report); the last SR carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set to <Display Name> (or Anonymous if user blocks caller ID), and TOOL is set to the Vendor/Hardware-platform-software-version. The NTP timestamp used in the SR is a snapshot of the local time for the ATA, not the time reported by an NTP server. If the ATA receives a RR from the peer, it attempts to compute the round trip delay and show it as the Call Round Trip Delay value (ms) on the Information page. Default setting: 0
No UDP Checksum
Select yes if you want the ATA to calculate the UDP header checksum for SIP messages. Otherwise, select no. Default setting: no
Stats In BYE
Determines whether the ATA includes the P-RTP-Stat header or response in a BYE message. The header contains the RTP statistics of the current call. Select yes or no from the drop-down list. Default setting: yes
The format of the P-RTP-Stat header is:
P-RTP-State: PS=<packets sent>,OS=<octets sent>,PR=<packets received>,OR=<octets received>,PL=<packets lost>,JI=<jitter in ms>,LA=<delay in ms>,DU=<call duration ins>,EN=<encoder>,DE=<decoder>.
SDP Payload Types
Field
Description
NSE Dynamic Payload
NSE dynamic payload type. The valid range is 96-127. Default setting: 100
AVT Dynamic Payload
AVT dynamic payload type. The valid range is 96-127. Default setting: 101
RTP-Start-Loopback Codec. Select one of the following: G711u, G711a, G726-32, G729a. Default setting: G711u
NSE Codec Name
NSE codec name used in SDP. Default setting: NSE
AVT Codec Name
AVT codec name used in SDP. Default setting: telephone-event
G711u Codec Name
G.711u codec name used in SDP. Default setting: PCMU
G711a Codec Name
G.711a codec name used in SDP. Default setting: PCMA
G726r32 Codec Name
G.726-32 codec name used in SDP. Default setting: G726-32
G729a Codec Name
G.729a codec name used in SDP. Default setting: G729a
G729b Codec Name
G.729b codec name used in SDP. Default setting: G729ab
EncapRTP Codec Name
EncapRTP codec name used in SDP. Default setting: encaprtp
NAT Support Parameters
Field
Description
Handle VIA received
If you select yes, the ATA processes the received parameter in the VIA header (this value is inserted by the server in a response to any one of its requests) If you select no, the parameter is ignored. Select yes or no from the drop-down menu. Default setting: no
Handle VIA rport
If you select yes, the ATA processes the rport parameter in the VIA header (this value is inserted by the server in a response to any one of its requests) If you select no, the parameter is ignored. Select yes or no from the drop-down menu. Default setting: no
Insert VIA received
Inserts the received parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu. Default setting: no
Insert VIA rport
Inserts the parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ. Select yes or no from the drop-down menu. Default setting: no
Substitute VIA Addr
Lets you use NAT-mapped IP:port values in the VIA header. Select yes or no from the drop-down menu. Default setting: no
Send Resp To Src Port
Sends responses to the request source port instead of the VIA sent-by port. Select yes or no from the drop-down menu. Default setting: no
STUN Enable
Enables the use of STUN to discover NAT mapping. Select yes or no from the drop-down menu. Default setting: no
STUN Test Enable
If the STUN Enable feature is enabled and a valid STUN server is available, the ATA can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a Warning header in all subsequent REGISTER requests. If the ATA detects symmetric NAT or a symmetric firewall, NAT mapping is disabled. Default setting: no
STUN Server
IP address or fully-qualified domain name of the STUN server to contact for NAT mapping discovery. Default setting: blank
EXT IP
External IP address to substitute for the actual IP address of the ATA in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.
If this parameter is specified, the ATA assumes this IP address when generating SIP messages and SDP (if NAT Mapping is enabled for that line) However, the results of STUN and VIA received parameter processing, if available, supersede this statically configured value.
This option requires that you have (1) a static IP address from your Internet Service Provider and (2) an edge device with a symmetric NAT mechanism. If the ATA is the edge device, the second requirement is met. Default setting: blank
EXT RTP Port Min
External port mapping number of the RTP Port Min. number. If this value is not zero, the RTP port number in all outgoing SIP messages is substituted for the corresponding port value in the external RTP port range. Default setting: blank
NAT Keep Alive Intvl
Interval between NAT-mapping keep alive messages. Default setting: 15
Redirect Keep Alive
Interval between NAT Redirect keep alive messages. Default setting: 15
Linksys Key System Parameters
Field
Description
Linksys Key System
To enable operation with the Cisco SPA9000, choose yes. Otherwise, choose no. Default setting: no
Multicast Address
The multicast address for devices in the Cisco SPA9000 voice network. Default setting: 224.168.168.168:6061
Key System Auto Discovery
To enable auto-discovery of the Cisco SPA9000 voice system, choose yes. Otherwise, choose no. Default setting: yes
Key System IP Address
The IP address of the Cisco SPA9000. Default setting: blank
Force LAN Codec
If needed, specify a voice codec. Default setting: none
Provisioning
Use the Voice > Provisioning page to configure profiles and parameters to provision the ATA from a remote server.
NOTE Cisco SPA122, SPA112, and SPA232D supports 302/310 redirect while provisioning.
To open this page: Click Voice on the menu bar, and then click Provisioning in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
Configuration Profile
Field
Description
Provision Enable
Controls all resync actions independently of firmware upgrade actions. Set to yes to enable remote provisioning. Default setting: yes
Resync On Reset
Triggers a resync after every reboot except for reboots caused by parameter updates and firmware upgrades. Default setting: yes
Resync Random Delay
The maximum value for a random time interval that the ATA waits before making its initial contact with the provisioning server. This delay is effective only on the initial configuration attempt following power-on or reset. The delay is a pseudo-random number between zero and this value.
This parameter is in units of 20 seconds; the default value of 2 represents 40 seconds. This feature is disabled when this parameter is set to zero.
This feature can be used to prevent an overload of the provisioning server when a large number of devices power-on simultaneously. Default setting: 2 (40 seconds)
Resync At (HHmm)
The time of day when the device tries to resync. The resync is performed each day. Used in conjunction with the Resync At Random Delay. Default setting: blank
Resync At Random Delay
Used in conjunction with the Resync At (HHmm) setting, this parameter sets a range of possible values for the resync delay. The system randomly chooses a value from this range and waits the specified number of seconds before attempting to resync. This feature is intended to prevent the network jam that would occur if all resynchronizing devices began the resync at the exact same time of day. Default setting: 600
Resync Periodic
The time interval between periodic resyncs with the provisioning server. The associated resync timer is active only after the first successful synchronization with the server. Setting this parameter to zero disables periodic resynchronization. Default setting:3600 seconds
Resync Error Retry Delay
Resync retry interval (in seconds) applied in case of resync failure.
The ATA has an error retry timer that activates if the previous attempt to sync with the provisioning server fails. The ATA waits to contact the server again until the timer counts down to zero.
This parameter is the value that is initially loaded into the error retry timer. If this parameter is set to zero, the ATA immediately retries to sync with the provisioning server following a failed attempt. Default setting:3600 seconds
Forced Resync Delay
Maximum delay (in seconds) that the ATA waits before performing a resync.
The ATA does not resync while one of its lines is active. Because a resync can take several seconds, it is desirable to wait until the ATA has been idle for an extended period before resynchronizing. This allows a user to make calls in succession without interruption.
The ATA has a timer that begins counting down when all of its lines become idle. This parameter is the initial value of the counter. Resync events are delayed until this counter decrements to zero. Default setting: 14400 seconds
Resync From SIP
Enables a resync to be triggered via a SIP NOTIFY message. Default setting: yes
Resync After Upgrade Attempt
Triggers a resync after every firmware upgrade attempt. Default setting: yes
Resync Trigger 1
Resync Trigger 2
Configurable resync trigger conditions. A resync is triggered when the logic equation in these parameters evaluates to TRUE. Default setting:blank
Resync Fails On FNF
Determines whether a file-not-found response from the provisioning server constitutes a successful or a failed resync. A failed resync activates the error resync timer. Default setting: yes
Profile Rule
This parameter is a profile script that evaluates to the provisioning resync command. The command is a TCP/IP operation and an associated URL. The TCP/IP operation can be TFTP, HTTP, or HTTPS.
If the command is not specified, TFTP is assumed, and the address of the TFTP server is obtained through DHCP option 66. In the URL, either the IP address or the FQDN of the server can be specified. The file name can have macros, such as $MA, which expands to the ATA MAC address. Default setting: /spa$PSN.cfg
Profile Rule B:
Profile Rule C:
Profile Rule D
Defines second, third, and fourth resync commands and associated profile URLs. These profile scripts are executed sequentially after the primary Profile Rule resync operation has completed. If a resync is triggered and Profile Rule is blank, Profile Rule B, C, and D are still evaluated and executed. Default setting: blank
Log Resync Request Msg
This parameter contains the message that is sent to the Syslog server at the start of a resync attempt. Default setting: $PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH
Log Resync Success Msg
Syslog message issued upon successful completion of a resync attempt. Default setting: $PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH
Log Resync Failure Msg
Syslog message issued after a failed resync attempt. Default setting: $PN $MAC -- Resync failed: $ERR
Report Rule
The target URL to which configuration reports are sent. This parameter has the same syntax as the Profile_Rule parameter, and resolves to a TCP/IP command with an associated URL.
A configuration report is generated in response to an authenticated SIP NOTIFY message, with Event: report. The report is an XML file containing the name and value of all the device parameters.
This parameter may optionally contain an encryption key. For example:
Determines whether or not firmware upgrade operations can occur independently of resync actions. Default setting: yes
Upgrade Error Retry Delay
The upgrade retry interval (in seconds) applied in case of upgrade failure. The ATA has a firmware upgrade error timer that activates after a failed firmware upgrade attempt. The timer is initialized with the value in this parameter. The next firmware upgrade attempt occurs when this timer counts down to zero. Default setting: 3600 seconds
Downgrade Rev Limit
Enforces a lower limit on the acceptable version number during a firmware upgrade or downgrade. The ATA does not complete a firmware upgrade operation unless the firmware version is greater than or equal to this parameter. Default setting:blank
Upgrade Rule
This parameter is a firmware upgrade script with the same syntax as Profile_Rule. Defines upgrade conditions and associated firmware URLs. Default setting:blank
Log Upgrade Request Msg
Syslog message issued at the start of a firmware upgrade attempt. Default setting: $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH
Log Upgrade Success Msg
Syslog message issued after a firmware upgrade attempt completes successfully. Default setting: $PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR
Log Upgrade Failure Msg
Syslog message issued after a failed firmware upgrade attempt. Default setting: $PN $MAC -- Upgrade failed: $ERR
License Keys
This field is not currently used.
CA Settings
Field
Description
Custom CA URL
The URL of a file location for a custom Certificate Authority (CA) certificate. Either the IP address or the FQDN of the server can be specified. The file name can have macros, such as $MA, which expands to the ATA MAC address. Default setting: null
General Purpose Parameters
Field
Description
GPP A to GPP P
General purpose provisioning parameters. These parameters can be used as variables in provisioning and upgrade rules. They are referenced by prepending the variable name with a `$' character, such as $GPP_A. Default setting: blank
Regional
Use the Voice > Regional page to localize your system with the appropriate regional settings.
To open this page: Click Voice on the menu bar, and then click Region in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
Defining Ring and Cadence and Tone Scripts
To define ring and tone patterns, the ATA uses the concept of scripts. Below is information about creating Cadence Scripts (CadScripts), Frequency Scripts (FreqScripts), and Tone Scripts (ToneScripts).
NOTE Total tone length is not configurable.
CadScript
A mini-script of up to 127 characters that specifies the cadence parameters of a signal.
Syntax: S1[;S2], where: Si=Di(oni,1/offi,1[,oni,2/offi,2[,oni,3/offi,3[,oni,4/offi,4[,oni,5/offi,5,oni,6/offi,6]]]]]) and is known as a section, oni,j and offi,j are the on/off duration in seconds of a segment and i = 1 or 2, and j = 1 to 6. Di is the total duration of the section in seconds. All durations can have up to three decimal places to provide 1 ms resolution. The wildcard character "*" represents infinite duration. The segments within a section are played in order and repeated until the total duration is played.
Example 1: 60(2/4)
Number of Cadence Sections = 1
Cadence Section 1: Section Length = 60 s
Number of Segments = 1
Segment 1: On=2s, Off=4s
Total Ring Length = 60s
Example 2—Distinctive ring (short,short,short,long): 60(.2/.2,.2/.2,.2/.2,1/4)
Number of Cadence Sections = 1
Cadence Section 1: Section Length = 60s
Number of Segments = 4
Segment 1: On=0.2s, Off=0.2s
Segment 2: On=0.2s, Off=0.2s
Segment 3: On=0.2s, Off=0.2s
Segment 4: On=1.0s, Off=4.0s
Total Ring Length = 60s
FreqScript
A mini-script of up to 127 characters that specifics the frequency and level parameters of a tone.
Syntax: F1@L1[,F2@L2[,F3@L3[,F4@L4[,F5@L5[,F6@L6]]]] Where F1-F6are frequency in Hz (unsigned integers only) and L1-L6are corresponding levels in dBm (with up to 1 decimal places) White spaces before and after the comma are allowed (but not recommended)
Example 1—Call Waiting Tone: 440@-10
Number of Frequencies = 1
Frequency 1 = 440 Hz at -10 dBm
Example 2—Dial Tone: 350@-19,440@-19
Number of Frequencies = 2
Frequency 1 = 350 Hz at -19 dBm
Frequency 2 = 440 Hz at -19 dBm
ToneScript
A mini-script of up to 127 characters that specifies the frequency, level and cadence parameters of a call progress tone. May contain up to 127 characters.
Syntax: FreqScript;Z1[;Z2]. The section Z1is similar to the S1section in a CadScript except that each on/off segment is followed by a frequency components parameter: Z1= D1(oni,1/offi,1/fi,1[,oni,2/offi,2/fi,2[,oni,3/offi,3/fi,3[,oni,4/offi,4/fi,4[,oni,5/offi,5/fi,5[,oni,6/offi,6/fi,6]]]]]), where fi,j = n1[+n2]+n3[+n4[+n5[+n6]]]]] and 1 < nk< 6 indicates which of the frequency components given in the FreqScript are used in that segment; if more than one frequency component is used in a segment, the components are summed together.
Example 1—Dial tone: 350@-19,440@-19;10(*/0/1+2)
Number of Frequencies = 2
Frequency 1 = 350 Hz at -19 dBm
Frequency 2 = 440 Hz at -19 dBm
Number of Cadence Sections = 1
Cadence Section 1: Section Length = 10 s
Number of Segments = 1
Segment 1: On=forever, with Frequencies 1 and 2
Total Tone Length = 10s
Example 2—Stutter tone: 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)
Number of Frequencies = 2
Frequency 1 = 350 Hz at -19 dBm
Frequency 2 = 440 Hz at -19 dBm
Number of Cadence Sections = 2
Cadence Section 1: Section Length = 2s
Number of Segments = 1
Segment 1: On=0.1s, Off=0.1s with Frequencies 1 and 2
Cadence Section 2: Section Length = 10s
Number of Segments = 1
Segment 1: On=forever, with Frequencies 1 and 2
Total Tone Length = 12s
Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
Call ProgressTones
Field
Description
Dial Tone
Prompts the user to enter a phone number. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out. Default setting: 350@-5,440@-5;10(*/0/1+2)
Second Dial Tone
Alternative to the Dial Tone when the user dials a three-way call. Default setting: 420@-5,520@-5;10(*/0/1+2)
Outside Dial Tone
Alternative to the Dial Tone. It prompts the user to enter an external phone number, as opposed to an internal extension. It is triggered by a comma character encountered in the dial plan. Default setting: 420@-4;10(*/0/1)
Prompt Tone
Prompts the user to enter a call forwarding phone number. Default setting: 520@-5,620@-5;10(*/0/1+2)
Busy Tone
Played when a 486 RSC is received for an outbound call. Default setting: 480@-5,620@-5;10(.5/.5/1+2)
Reorder Tone
Played when an outbound call has failed, or after the far end hangs up during an established call. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out. Default setting: 480@-5,620@-5;10(.25/.25/1+2)
Off Hook Warning Tone
Played when the caller has not properly placed the handset on the cradle. Off Hook Warning Tone is played when the Reorder Tone times out. Default setting: 480@-3,620@3;10(.125/.125/1+2)
Ring Back Tone
Played during an outbound call when the far end is ringing. Default setting: 440@-5,480@-5;*(2/4/1+2)
Ring Back 2 Tone
Your ATA plays this ringback tone instead of Ring Back Tone if the called party replies with a SIP 182 response without SDP to its outbound INVITE request. Default setting: the same as Ring Back Tone, except the cadence is 1s on and 1s off. Default setting: 440@-5,480@-5;*(1/1/1+2)
Confirm Tone
Brief tone to notify the user that the last input value has been accepted. Default setting: 600@-4;1(.25/.25/1)
SIT1 Tone
Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. Default setting: 985@-4,1428@-4,1777@-4;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)
SIT2 Tone
Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. Default setting: 914@-4,1371@-4,1777@-4;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)
SIT3 Tone
Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. Default setting: 914@-4,1371@-4,1777@-4;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)
SIT4 Tone
Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen. Default setting: 985@-4,1371@-4,1777@-4;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)
MWI Dial Tone
Played instead of the Dial Tone when there are unheard messages in the caller's mailbox. Default setting: 350@-5,440@-5;2(.1/.1/1+2);10(*/0/1+2)
Cfwd Dial Tone
Played when all calls are forwarded. Default setting: 350@-5,440@-5;2(.2/.2/1+2);10(*/0/1+2)
Holding Tone
Informs the local caller that the far end has placed the call on hold. Default setting: 600@-5;*(.1/.1/1,.1/.1/1,.1/9.5/1)
Conference Tone
Played to all parties when a three-way conference call is in progress. Default setting: 350@-5;20(.1/.1/1,.1/9.7/1)
Secure Call Indication Tone
Played when a call has been successfully switched to secure mode. It should be played only for a short while (less than 30 seconds) and at a reduced level (less than -19 dBm) so it does not interfere with the conversation. Default setting: 397@-5,507@-5;15(0/2/0,.2/.1/1,.1/2.1/2)
VoIP PIN Tone
This tone is played to prompt a VoIP caller to enter a PIN number.
PSTN PIN Tone
This tone is played to prompt a PSTN caller to enter a PIN number.
Feature Invocation Tone
Played when a feature is implemented. Default setting: 350@-4;*(.1/.1/1)
Call Remind Tone
When there are 2 calls on the FXS port, and if one call is held by the UUT (Unit under test)and the other is connected, a holding tone is played on the UUT to remind the presence of the held call.
Default setting: blank (the feature is not enabled)
Distinctive Ring Patterns
Field
Description
Ring1 Cadence
Cadence script for distinctive ring 1. Default setting: 60(2/4)
Ring2 Cadence
Cadence script for distinctive ring 2. Default setting: 60(.8/.4,.8/4)
Ring3 Cadence
Cadence script for distinctive ring 3. Default setting: 60(.4/.2,.4/.2,.8/4)
Ring4 Cadence
Cadence script for distinctive ring 4. Default setting: 60(.3/.2,1/.2,.3/4)
Ring5 Cadence
Cadence script for distinctive ring 5. Default setting: 1(.5/.5)
Ring6 Cadence
Cadence script for distinctive ring 6. Default setting: 60(.2/.4,.2/.4,.2/4)
Ring7 Cadence
Cadence script for distinctive ring 7. Default setting: 60(.4/.2,.4/.2,.4/4)
Ring8 Cadence
Cadence script for distinctive ring 8. Default setting: 60(0.25/9.75)
Distinctive Call Waiting Tone Patterns
Field
Description
CWT1 Cadence
Cadence script for distinctive CWT 1. Default setting: 30(.3/9.7)
CWT2 Cadence
Cadence script for distinctive CWT 2. Default setting: 30(.1/.1, .1/9.7)
Cadence script for distinctive CWT 5. Default setting: 1(.5/.5)
CWT6 Cadence
Cadence script for distinctive CWT 6. Default setting: 30(.3/.1,.3/.1,.1/9.1)
CWT7 Cadence
Cadence script for distinctive CWT 7. Default setting: 30(.3/.1,.3/.1,.1/9.1)
CWT8 Cadence
Cadence script for distinctive CWT 8. Default setting: 2.3(.3/2)
Distinctive Ring/CWT Pattern Names
Field
Description
Ring1 Name
Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 1 for the inbound call. Default setting: Bellcore-r1
Ring2 Name
Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 2 for the inbound call. Default setting: Bellcore-r2
Ring3 Name
Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 3 for the inbound call. Default setting: Bellcore-r3
Ring4 Name
Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 4 for the inbound call. Default setting: Bellcore-r4
Ring5 Name
Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 5 for the inbound call. Default setting: Bellcore-r5
Ring6 Name
Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 6 for the inbound call. Default setting: Bellcore-r6
Ring7 Name
Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 7 for the inbound call. Default setting: Bellcore-r7
Ring8 Name
Name in an INVITE's Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call. Default setting: Bellcore-r8
Ring and Call Waiting Tone Spec
IMPORTANT: Ring and Call Waiting tones do not work the same way on all phones. When setting ring tones, consider the following recommendations:
•Begin with the default Ring Waveform, Ring Frequency, and Ring Voltage.
•If your ring cadence doesn't sound right, or your phone doesn't ring, change the following settings:
–Ring Waveform: Sinusoid
–Ring Frequency: 25
–Ring Voltage: 80Vc
Field
Description
Ring Waveform
Waveform for the ringing signal. Choices are Sinusoidor Trapezoid. Default setting: Sinusoid
Ring Frequency
Frequency of the ringing signal. Valid values are 10-100 (Hz) Default setting: 20
Ring Voltage
Ringing voltage. Choices are 60-90 (V) Default setting: 85
CWT Frequency
Frequency script of the call waiting tone. All distinctive CWTs are based on this tone. Default setting: 440@-10
Synchronized Ring
If this is set to yes, when the ATA is called, all lines ring at the same time (similar to a regular PSTN line) After one line answers, the others stop ringing. Default setting: no
Control Timer Values (sec)
Field
Description
Hook Flash Timer Min
Minimum on-hook time before off-hook qualifies as hook flash. Less than this the on-hook event is ignored. Range: 0.1-0.4 seconds. Default setting: 0.1
Hook Flash Timer Max
Maximum on-hook time before off-hook qualifies as hook flash. More than this the on-hook event is treated as on hook (no hook-flash event) Range: 0.4-1.6 seconds. Default setting: 0.9
Callee On Hook Delay
Phone must be on-hook for at this time in sec. before the ATA will tear down the current inbound call. It does not apply to outbound calls. Range: 0-255 seconds. Default setting: 0
Reorder Delay
Delay after far end hangs up before reorder tone is played. 0 = plays immediately, inf = never plays. Range: 0-255 seconds. Default setting: 5.
Call Back Expires
Expiration time in seconds of a call back activation. Range: 0-65535 seconds. Default setting: 1800
Call Back Retry Intvl
Call back retry interval in seconds. Range: 0-255 seconds. Default setting: 30
Call Back Delay
Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the ATA still considers the call as failed and keeps on retrying. Default setting: 0.5
VMWI Refresh Intvl
Interval between VMWI refresh to the device. Default setting: 0
Interdigit Long Timer
Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed. Range: 0-64 seconds. Default setting: 10
Interdigit Short Timer
Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences. Range: 0-64 seconds. Default setting: 3
CPC Delay
Delay in seconds after caller hangs up when the ATA starts removing the tip-and-ring voltage to the attached equipment of the called party. The range is 0-255 seconds. This feature is generally used for answer supervision on the caller side to signal to the attached equipment when the call has been connected (remote end has answered) or disconnected (remote end has hung up) This feature should be disabled for the called party (in other words, by using the same polarity for connected and idle state) and the CPC feature should be used instead.
Without CPC enabled, reorder tone will is played after a configurable delay. If CPC is enabled, dial tone will be played when tip-to-ring voltage is restored. Resolution is 1 second. Default setting: 2
CPC Duration
Duration in seconds for which the tip-to-ring voltage is removed after the caller hangs up. After that, tip-to-ring voltage is restored and the dial tone applies if the attached equipment is still off-hook. CPC is disabled if this value is set to 0. Range: 0 to 1.000 second. Resolution is 0.001 second. Default setting: 0(CPC disabled)
Vertical Service Activation Codes
Vertical Service Activation Codes are automatically appended to the dial-plan. There is no need to include them in dial-plan, although no harm is done if they are inclu`ded.
Field
Description
Call Return Code
Call Return Code This code calls the last caller. Default setting: *69
Call Redial Code
Redials the last number called. Default setting: *07
Blind Transfer Code
Begins a blind transfer of the current call to the extension specified after the activation code. Default setting: *98
Call Back Act Code
Starts a callback when the last outbound call is not busy. Default setting: *66
Call Back Deact Code
Cancels a callback. Default setting: *86
Call Back Busy Act Code
Starts a callback when the last outbound call is busy. Default setting: *05
Cfwd All Act Code
Forwards all calls to the extension specified after the activation code. Default setting: *72
Cfwd All Deact Code
Cancels call forwarding of all calls. Default setting: *73
Cfwd Busy Act Code
Forwards busy calls to the extension specified after the activation code. Default setting: *90
Cfwd Busy Deact Code
Cancels call forwarding of busy calls. Default setting: *91
Cfwd No Ans Act Code
Forwards no-answer calls to the extension specified after the activation code. Default setting: *92
Cfwd No Ans Deact Code
Cancels call forwarding of no-answer calls. Default setting: *93
Cfwd Last Act Code
Forwards the last inbound or outbound call to the number that the user specifies after entering the activation code. Default setting: *63
Cfwd Last Deact Code
Cancels call forwarding of the last inbound or outbound call. Default setting: *83
Block Last Act Code
Blocks the last inbound call. Default setting: *60
Block Last Deact Code
Cancels blocking of the last inbound call. Default setting: *80
Accept Last Act Code
Accepts the last outbound call. It lets the call ring through when do not disturb or call forwarding of all calls are enabled. Default setting: *64
Accept Last Deact Code
Cancels the code to accept the last outbound call. Default setting: *84
CW Act Code
Enables call waiting on all calls. Default setting: *56
CW Deact Code
Disables call waiting on all calls. Default setting: *57
CW Per Call Act Code
Enables call waiting for the next call. Default setting: *71
CW Per Call Deact Code
Disables call waiting for the next call. Default setting: *70
Block CID Act Code
Blocks caller ID on all outbound calls. Default setting: *67
Block CID Deact Code
Removes caller ID blocking on all outbound calls. Default setting: *68
Block CID Per Call Act Code
Blocks caller ID on the next outbound call. Default setting: *81
Block CID Per Call Deact Code
Removes caller ID blocking on the next inbound call. Default setting: *82
Block ANC Act Code
Blocks all anonymous calls. Default setting: *77
Block ANC Deact Code
Removes blocking of all anonymous calls. Default setting: *87
DND Act Code
Enables the do not disturb feature. Default setting: *78
DND Deact Code
Disables the do not disturb feature. Default setting: *79
CID Act Code
Enables caller ID generation. Default setting: *65
CID Deact Code
Disables caller ID generation. Default setting: *85
CWCID Act Code
Enables call waiting, caller ID generation. Default setting: *25
CWCID Deact Code
Disables call waiting, caller ID generation. Default setting: *45
Dist Ring Act Code
Enables the distinctive ringing feature. Default setting: *26
Dist Ring Deact Code
Disables the distinctive ringing feature. Default setting: *46
Speed Dial Act Code
Assigns a speed dial number. Default setting: *74
Paging Code
Used for paging other clients in the group. Default setting: *96
Secure All Call Act Code
Makes all outbound calls secure. Default setting: *16
Secure No Call Act Code
Makes all outbound calls not secure. Default setting: *17
Secure One Call Act Code
Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.) Default setting: *18
Secure One Call Deact Code
Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default.) Default setting: *19
Conference Act Code
If this code is specified, the user must enter it before dialing the third party for a conference call. Enter the code for a conference call. Default setting: blank
Attn-Xfer Act Code
If the code is specified, the user must enter it before dialing the third party for a call transfer. Enter the code for a call transfer. Default setting: blank
Modem Line Toggle Code
Toggles the line to a modem. Modem passthrough mode can be triggered only by pre-dialing this code. Default setting: *99
FAX Line Toggle Code
Toggles the line to a fax machine. Default setting: #99
Media Loopback Code
Use for media loopback. Default setting: *03
Referral Services Codes
These codes tell the ATA what to do when the user places the current call on hold and is listening to the second dial tone.
One or more *codes can be configured into this parameter, such as *98, or *97|*98|*123, etc. The maximum length is 79 characters. This parameter applies when the user places the current call on hold by pressing the hook flash button. Each *code (and the following valid target number according to current dial plan) triggers the ATA to perform a blind transfer to a target number that is prepended by the service *code.
For example, after the user dials *98, the ATA plays a special dial tone called the Prompt Tone while waiting for the user the enter a target number (which is checked according to dial plan as in normal dialing). When a complete number is entered, the ATA sends a blind REFER to the holding party with the Refer-To target equal to *98 target_number. This feature allows the ATA to hand off a call to an application server to perform further processing, such as call park.
The *codes should not conflict with any of the other vertical service codes internally processed by the ATA. You can empty the corresponding *code that you do not want the ATA to process. Default setting: blank
Feature Dial Services Codes
These codes tell the ATA what to do when the user is listening to the first or second dial tone.
One or more *codes can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. The maximum length is 79 characters. This parameter applies when the user has a dial tone (first or second dial tone) After receiving dial tone, a user enters the *code and the target number according to current dial plan. For example, after user dials *72, the ATA plays a special tone called a Prompt tone while awaiting the user to enter a valid target number. When a complete number is entered, the ATA sends a INVITE to *72 target_number as in a normal call. This feature allows the proxy to process features like call forward (*72) or Block Caller ID (*67)
The *codes should not conflict with any of the other vertical service codes internally processed by the ATA. You can remove a corresponding *code that you do not want to the ATA to process.
You can add a parameter to indicate which tone plays after the *code is entered, such as *72`c`|*67`p`. Below is a list of allowed tone parameters (note the use of open quotes surrounding the parameter, without spaces)
`c` = <Cfwd Dial Tone> `d` = <Dial Tone> `m` = <MWI Dial Tone> `o` = <Outside Dial Tone> `p` = <Prompt Dial Tone> `s` = <Second Dial Tone> `x` = No tones are place, x is any digit not used above
If no tone parameter is specified, the ATA plays Prompt tone by default. If the *code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not include this parameter. Instead, add the *code in the dial plan and the ATA send INVITE *73@..... as usual when user dials *73. Default setting: blank
Vertical Service Announcement Codes
Field
Description
Service Annc Base Number
Base number for service announcements. Default setting: blank
Service Annc Extension Codes
Extension codes for service announcements. Default setting: blank
Outbound Call Codec Selection Codes
Field
Description
Prefer G711u Code
Dial prefix to make G.711u the preferred codec for the call. Default setting: *017110
Force G711u Code
Dial prefix to make G.711u the only codec that can be used for the call. Default setting: *027110
Prefer G711a Code
Dial prefix to make G.711a the preferred codec for the call. Default setting: *017111
Force G711a Code
Dial prefix to make G.711a the only codec that can be used for the call. Default setting: *027111
Prefer G726r32 Code
Dial prefix to make G.726r32 the preferred codec for the call. Default setting: *0172632
Force G726r32 Code
Dial prefix to make G.726r32 the only codec that can be used for the call. Default setting: *0272632
Prefer G729a Code
Dial prefix to make G.729a the preferred codec for the call. Default setting: *01729
Force G729a Code
Dial prefix to make G.729a the only codec that can be used for the call. Default setting: *02729
Prefer G722 Code
Dial prefix to make G.722 the preferred codec for the call. Default setting: *01722
Force G722 Code
Dial prefix to make G.722 the only codec that can be used for the call. Default setting: *02722
Miscellaneous
Field
Description
FXS Port Impedance
Sets the electrical impedance of the PHONE port. Choices are:
NOTE For New Zealand impedance (370+620||310nF), use 270+750||150nF.
FXS Port Input Gain
Input gain in dB, up to three decimal places. The range is 6.000 to -12.000. Default setting: -3.
FXS Port Output Gain
Output gain in dB, up to three decimal places. The range is 6.000 to -12.000. The Call Progress Tones and DTMF playback level are not affected by the FXS Port Output Gain parameter. Default setting: -3.
DTMF Playback Level
Local DTMF playback level in dBm, up to one decimal place. Default setting: -16.0.
DTMF Twist
To gain difference between the two tone frequency.
Default setting: 2
DTMF Playback Length
Local DTMF playback duration in milliseconds. Default setting: .1.
Detect ABCD
To enable local detection of DTMF ABCD, select yes. Otherwise, select no. Default setting: yes
This setting has no effect if DTMF Tx Method is INFO; ABCD is always sent OOB regardless in this setting.
Playback ABCD
To enable local playback of OOB DTMF ABCD, select yes. Otherwise, select no. Default setting: yes
Caller ID Method
The choices are described below. Default setting: Bellcore(N.Amer, China)
•Bellcore (N.Amer,China): CID, CIDCW, and VMWI. FSK sent after first ring (same as ETSI FSK sent after first ring) (no polarity reversal or DTAS)
•DTMF (Finland, Sweden): CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.
•DTMF (Denmark): CID only. DTMF sent before first ring with no polarity reversal and no DTAS.
•ETSI DTMF: CID only. DTMF sent after DTAS (and no polarity reversal) and before first ring.
•ETSI DTMF With PR: CID only. DTMF sent after polarity reversal and DTAS and before first ring.
•ETSI DTMF After Ring: CID only. DTMF sent after first ring (no polarity reversal or DTAS)
•ETSI FSK: CID, CIDCW, and VMWI. FSK sent after DTAS (but no polarity reversal) and before first ring. Waits for ACK from a device after DTAS for CIDCW.
•ETSI FSK With PR (UK): CID, CIDCW, and VMWI. FSK is sent after polarity reversal and DTAS and before first ring. Waits for ACK from a device after DTAS for CIDCW. Polarity reversal is applied only if equipment is on hook.
•DTMF (Denmark) with PR: CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring. Default setting: Bellcore(N.Amer, China)
FXS Port Power Limit
The choices are from 1 to 8. Default setting: 3
Caller ID FSK Standard
The ATA supports bell 202 and v.23 standards for caller ID generation. Default setting:bell 202
Feature Invocation Method
Select the method you want to use, Default or Sweden default. Default setting: Default
Line 1 Settings (PHONE Port)
Use the Voice > Line 1 page to configure the settings for calls through the PHONE port.
To open this page: Click Voice on the menu bar, and then click Line 1 in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
NOTE In a configuration profile, the FXS parameters must be appended with the appropriate numeral (for example, [1] or [2]) to identify the port to which the setting applies.
Line Enable
Field
Description
Line Enable
To enable this line for service, select yes. Otherwise, select no. Default setting: yes
Streaming Audio Server (SAS)
Field
Description
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 setting: 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 ATA ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled) Default setting: 30
SAS Inbound RTP Sink
The purpose of this parameter is to work around devices that do not play inbound RTP if the SAS line declares itself as a send-only device and tells the client not to stream out audio. This parameter is an FQDN or IP address of an RTP sink to be used by the 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, if specified, will appear in the m = line of the SDP. If this value is not specified or is equal to 0, then c = 0.0.0.0 and a=sendonly will be 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 setting: blank
NAT Settings
Field
Description
NAT Mapping Enable
To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. Default setting: no
NAT Keep Alive Enable
To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. Default setting: no
NAT Keep Alive Msg
Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent. Default setting: $NOTIFY
NAT Keep Alive Dest
Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current proxy server or outbound proxy server. Default setting: $PROXY
Network Settings
Field
Description
SIP ToS/DiffServ Value
TOS/DiffServ field value in UDP IP packets carrying a SIP message. Default setting: 0x68
SIP CoS Value
CoS value for SIP messages. Valid values are 0 through 7. Default setting: 3
RTP ToS/DiffServ Value
ToS/DiffServ field value in UDP IP packets carrying RTP data. Default setting: 0xb8
RTP CoS Value [0- 7]
CoS value for RTP data. Valid values are 0 through 7. Default setting: 6
Network Jitter Level
Determines how jitter buffer size is adjusted by the ATA. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high. Default setting: high
Jitter Buffer Adjustment
Choose yes to enable or no to disable this feature. Default setting: yes
SIP Settings
Field
Description
SIP Transport
The TCP choice provides "guaranteed delivery", which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. As a result, TCP overcomes the main disadvantages of UDP. In addition, for security reasons, most corporate firewalls block UDP ports. With TCP, new ports do not need to be opened or packets dropped, because TCP is already in use for basic activities such as Internet browsing or e-commerce. Options are: UDP, TCP, TLS. Default setting: UDP
SIP Port
Port number of the SIP message listening and transmission port. Default setting: 5060
SIP 100REL Enable
To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. Default setting: no
EXT SIP Port
The external SIP port number. Default setting: blank
Auth Resync-Reboot
If this feature is enabled, the ATA authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no. Default setting: yes
SIP Proxy-Require
The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported. Enter the appropriate header in the field provided. Default setting: blank
SIP Remote-Party-ID
To use the Remote-Party-ID header instead of the From header, select yes. Otherwise, select no. Default setting: yes
SIP GUID
This feature limits the registration of SIP accounts. The Global Unique ID is generated for each line for each ATA. When it is enabled, the ATA adds a GUID header in the SIP request. The GUID is generated the first time the unit boots up and stays with the unit through rebooting and even factory reset. Default setting: no
SIP Debug Option
SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. The choices are described below. Default setting: none
•none—No logging.
•1-line—Logs the start-line only for all messages.
•1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses.
•1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses.
•1-line excl. REG—Logs the start-line only for all messages except REGISTER requests/responses.
•1-line excl. OPT|NTFY|REG—Logs the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses.
•full—Logs all SIP messages in full text.
•full excl. OPT—Logs all SIP messages in full text except OPTIONS requests/responses.
•full excl. NTFY—Logs all SIP messages in full text except NOTIFY requests/responses.
•full excl. REG—Logs all SIP messages in full text except REGISTER requests/responses.
•full excl. OPT|NTFY|REG—Logs all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/responses.
RTP Log Intvl
The interval for the RTP log. Default setting: 0
Restrict Source IP
If configured, the ATA drops all packets sent to its SIP Ports from an untrusted IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured Proxy (or Outbound Proxy if Use Outbound Proxy is yes) Default setting: no
Referor Bye Delay
The number of seconds to wait before sending a BYE to the referrer to terminate a stale call leg after a call transfer.
Refer Target Bye Delay
The number of seconds to wait before sending a BYE to the refer target to terminate a stale call leg after a call transfer.
Referee Bye Delay
The number of seconds to wait before sending a BYE to the referee to terminate a stale call leg after a call transfer.
Refer-To Target Contact
To contact the refer-to target, select yes. Otherwise, select no. Default setting: no
Sticky 183
If this feature is enabled, the ATA 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 setting: no
Auth INVITE
When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy. Default setting: no
Reply 182 On Call Waiting
When enabled, the ATA replies with a SIP182 response to the caller if it is already in a call and the line is off-hook. To use this feature select yes. Default setting: no
Use Anonymous With RPID
Determines whether or not the ATA uses "Anonymous" when Remote Party ID is requested in the SIP message. Default setting: yes
Use Local Addr In From
Use the local ATA IP address in the SIP FROM message. Default setting: no
Call Feature Settings
Field
Description
Blind Attn-Xfer Enable
Enables the ATA 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 ATA 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 setting: 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. Default setting: blank
Xfer When Hangup Conf
Makes the ATA perform a transfer when a conference call has ended. Select yes or no from the drop-down menu. Default setting: 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. For example: conf@mysefver.com:12345 or conf (which uses the Proxy value as the domain). Default setting: blank
Conference Bridge Ports
Select the maximum number of conference call participants. The range is 3 to 10. Default setting: 3
NOTE When FXS port initiates a conference call, the caller ID on the analogue phone is updatedto the Conference Bridge.
Enable IP Dialing
Enable or disable IP dialing. If IP dialing is enabled, one can dial [userid@] a.b.c.d[:port], where `@', `.', and `:' are dialed by entering *, user-id must be numeric (like a phone number) and a, b, c, d must be between 0 and 255, and port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are optional. If the user-id portion matches a pattern in the dial plan, then it is interpreted as a regular phone number according to the dial plan. The INVITE message, however, is still sent to the outbound proxy if it is enabled. Default setting: no
Emergency Number
Comma separated list of emergency number patterns. If outbound call matches one of the pattern, the ATA will disable hook flash event handling. The condition is restored to normal after the call ends. Blank signifies that there is no emergency number. Maximum number length is 63 characters. Default setting: blank
Mailbox ID
Enter the ID number of the mailbox for this line. Default setting: blank
Proxy and Registration
Field
Description
Proxy
SIP proxy server for all outbound requests. Default setting: blank
Outbound Proxy
SIP Outbound Proxy Server where all outbound requests are sent as the first hop. Default setting: blank
Use Outbound Proxy
Enables the use of an Outbound Proxy. If set to no, the Outbound Proxy and Use OB Proxy in Dialog parameters are ignored. Default setting: no
Use OB Proxy In Dialog
Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the parameter Use Outbound Proxy is no, or the Outbound Proxy parameter is empty. Default setting: yes
Register
Enable periodic registration with the Proxy parameter. This parameter is ignored if Proxy is not specified. Default setting: yes
Make Call Without Reg
Allow making outbound calls without successful (dynamic) registration by the unit. If No, dial tone will not play unless registration is successful. Default setting: no
Register Expires
Expires value in sec in a REGISTER request. The ATA will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 - (231 - 1) sec. Default setting: 3600
Ans Call Without Reg
Allow answering inbound calls without successful (dynamic) registration by the unit. Default setting: no
Use DNS SRV
Whether to use DNS SRV lookup for Proxy and Outbound Proxy. Default setting: no
DNS SRV Auto Prefix
If enabled, the ATA will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name. Default setting: no
Proxy Fallback Intvl
After failing over to a lower priority server, the ATA waits for the specified Proxy Fallback Interval, in seconds, before retrying the highest priority proxy (or outbound proxy) servers. This parameter is useful only if the primary and backup proxy server list is provided to the ATA via DNS SRV record lookup on the server name. (Using multiple DNS A records per server name does not allow the notion of priority, so all hosts will be considered at the same priority and the ATA will not attempt to fall back after a failover.) Default setting: 3600
Proxy Redundancy Method
The method that the ATA uses to create a list of proxies returned in the DNS SRV records. If you select Normal, the list will contain proxies ranked by weight and priority. If you select Based on SRV port, the ATA also inspects the port number based on 1st proxy's port. Default setting: Normal
Mailbox Subscribe URL
The URL or IP address of the voicemail server. Default setting: blank
Mailbox Subscribe Expires
Sets subscription interval for voicemail message waiting indication. When this time period expires, the ATA sends another subscribe message to the voice mail server. Default: 2147483647
Subscriber Information
Field
Description
Display Name
Display name for caller ID. Default setting: blank
User ID
User ID for this line. Default setting: blank
Password
Password for this line. Default setting: blank
Use Auth ID
To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password. Default setting: no
Auth ID
Authentication ID for SIP authentication. Default setting: blank
Resident Online Number
This setting allows you to associate a "local" telephone number with this line using a valid Skype Online Number from Skype. Calls made to that number will ring your phone. Enter the number without spaces or special characters. Default setting: blank
Supplementary Service Subscription
The ATA provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service. A supplementary service should be disabled if a) the user has not subscribed for it, or b) the Service Provider intends to support similar service using other means than relying on the ATA.
Field
Description
Call Waiting Serv
Enable Call Waiting Service. Default setting: yes
Block CID Serv
Enable Block Caller ID Service. Default setting: yes
Block ANC Serv
Enable Block Anonymous Calls Service Default setting: yes
Dist Ring Serv
Enable Distinctive Ringing Service Default setting: yes
Cfwd All Serv
Enable Call Forward All Service Default setting: yes
Cfwd Busy Serv
Enable Call Forward Busy Service Default setting: yes
Cfwd No Ans Serv
Enable Call Forward No Answer Service Default setting: yes
Cfwd Sel Serv
Enable Call Forward Selective Service. Configure this service in the Selective Call Forward Settings section. Default setting: yes
Cfwd Last Serv
Enable Forward Last Call Service Default setting: yes
Block Last Serv
Enable Block Last Call Service Default setting: yes
Accept Last Serv
Enable Accept Last Call Service Default setting: yes
DND Serv
Enable Do Not Disturb Service Default setting: yes
CID-Serv
Enable Caller ID Service Default setting: yes
CWCID Serv
Enable Call Waiting Caller ID Service Default setting: yes
Call Return Serv
Enable Call Return Service Default setting: yes
Call Redial Serv
Enable Call Redial Service.
Call Back Serv
Enable Call Back Service.
Three Way Call Serv
Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer. Default setting: yes
Three Way Conf Serv
Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer. Default setting: yes
Attn Transfer Serv
Enable Attended Call Transfer Service. Three Way Conference is required for Attended Transfer. Default setting: yes
Unattn Transfer Serv
Enable Unattended (Blind) Call Transfer Service. Default setting: yes
MWI Serv
Enable MWI Service. MWI is available only if a Voice Mail Service is set-up in the deployment. Default setting: yes
VMWI Serv
Enable VMWI Service (FSK) Default setting: yes
Speed Dial Serv
Enable Speed Dial Service. Default setting: yes
Secure Call Serv
Secure Call Service. If this feature is enabled, a user can make a secure call by entering an activation code (*18 by default) before dialing the target number. Then audio traffic in both directions is encrypted for the duration of the call. Default setting: yes
Star codes are set in Vertical Service Activation Codes. To enable secure calling by default, without requiring a star code, set the user's Secure Call Setting to yes. See User 1.
Referral Serv
Enable Referral Service. See the Referral Services Codes parameter For more information. Default setting: yes
Feature Dial Serv
Enable Feature Dial Service. See the Feature Dial Services Codes parameter For more information. Default setting: yes
Service Announcement Serv
Enable Service Announcement Service. Default setting: no
Reuse CID Number As Name
Use the Caller ID number as the caller name. Default settings: yes
Audio Configuration
Field
Description
Preferred Codec, Second Preferred Codec, Third Preferred Codec
Up to three codecs to be used for all calls from this handset, listed order of preference. The actual codec used in a call still depends on the outcome of the codec negotiation protocol. Select one of the following: G711u, G711a, G726-32, G729a, or G722. Default setting for Preferred Codec: G711u Default setting for Second and Third Preferred Codec: Unspecified
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 setting: yes
Use Remote Pref Codec
To use the preferred codec specified by the remote peer, select yes. Otherwise, select no. Default setting:
Codec Negotiation
Specify the codecs for codec negotiation: Default or List All. Default setting: Default
G729a Enable
To enable the use of the G.729a codec at 8 kbps, select yes. Otherwise, select no. Default setting: yes
Silence Supp Enable
To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no. Default setting: no
G726-32 Enable
To enable the use of the G.726 codec at 32 kbps, select yes. Otherwise, select no. Default setting: yes
Silence Threshold
Select the appropriate setting for the threshold: high, medium, or low. Default setting: medium
FAX V21 Detect Enable
To enable detection of V21 fax tones, select yes. Otherwise, select no. Default setting: yes
Echo Canc Enable
To enable the use of the echo canceller, select yes. Otherwise, select no. Default setting: yes
FAX CNG Detect Enable
To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no. Default setting: yes
FAX Passthru Codec
Select the codec for fax passthrough, G711u or G711a. Default setting: G711u
Echo Canc Adapt Enable
To enable the echo canceller to adapt, select yes. Otherwise, select no. Default setting: yes
FAX Codec Symmetric
To force the ATA to use a symmetric codec during fax passthrough, select yes. Otherwise, select no. Default setting: yes
DTMF Process INFO
To use the DTMF process info feature, select yes. Otherwise, select no. Default setting: yes
FAX Passthru Method
Select the fax passthrough method: None, NSE, or ReINVITE. Default setting: NSE
DTMF Process AVT
To use the DTMF process AVT feature, select yes. Otherwise, select no. Default setting: yes
FAX Process NSE
To use the fax process NSE feature, select yes. Otherwise, select no. Default setting: yes
DTMF Tx Method
Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, or Auto. InBand sends DTMF by 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. Default setting: Auto
FAX Disable ECAN
If enabled, this feature automatically disables the echo canceller when a fax tone is detected. To use this feature, select yes. Otherwise, select no. Default setting: no
DTMF Tx Mode
DTMF Detection Tx Mode is available for SIP information and AVT. Options are: Strict or Normal. Default setting: Strict for which the following are true:
•A DTMF digit requires an extra hold time after detection.
•The DTMF level threshold is raised to -20 dBm.
The minimum and maximum duration thresholds are:
•strict mode for AVT: 70 ms
•normal mode for AVT: 40 ms
•strict mode for SIP info: 90 ms
•normal mode for SIP info: 50 ms
DTMF Tx Strict Hold Off Time
This parameter is in effect only when DTMF Tx Mode is set to strict, and when DTMF Tx Method is set to out-ofband; i.e. either AVT or SIP-INFO. The value can be set as low as 40 ms. There is no maximum limit. A larger value will reduce the chance of talk-off (beeping) during conversation, at the expense of reduced performance of DTMF detection, which is needed for interactive voice response systems (IVR) Default: 70 ms
FAX Enable T38
To enable the use of ITU-T T.38 standard for FAX Relay, select yes. Otherwise select no. Default setting: 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. Default setting: None
FAX T38 Redundancy
Select the appropriate number to indicate the number of previous packet payloads to repeat with each packet. Choose 0 for no payload redundancy. The higher the number, the larger the packet size and the more bandwidth consumed. Default setting: 1
•caller or callee: The ATA will detect FAX tone whether it is callee or caller
•caller only: The ATA will detect FAX tone only if it is the caller
•callee only: The ATA will detect FAX tone only if it is the callee Default setting: caller or callee.
Symmetric RTP
Enable symmetric RTP operation. If enabled, the ATA sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) the ATA sends RTP to the destination as indicated in the inbound SDP. Default setting: no
Fax T38 Return to Voice
When this feature is enabled, upon completion of the fax image transfer, the connection remains established and reverts to a voice call using the previously designated codec. Select yes to enable this feature, or select no to disable it. Default setting: no
Configuration Tables
G3 NSE-Based Passthrough FAX Call
Parameter
Value
Description
FAX V21 Detect Enable
Yes
FAX CNG Detect Enable
Yes
If CNG detection is not needed, set it to No.
FAX Tone Detect Mode
Caller or Callee
FAX Passthru Method
NSE
FAX Process NSE
Yes
FAX Passthru Codec
G.711u or G.711a
Depends on the passthrough codec used.
FAX Disable ECAN
No
Echo Canc Enable
Yes
FAX Enable T38
No
Modem Line
No
SG3/V.34 NSE-Based Passthrough FAX Call
Parameter
Value
Description
FAX V21 Detect Enable
Yes
FAX CNG Detect Enable
Yes
If CNG detection is not needed, set it to No.
FAX Tone Detect Mode
Caller or Callee
FAX Passthru Method
NSE
FAX Process NSE
Yes
FAX Passthru Codec
G.711u or G.711a
Depends on the passthrough codec used.
FAX Disable ECAN
No
Echo Canc Enable
Yes
FAX Enable T38
No
Modem Line
No
SIP Protocol-Based passthrough FAX Call
Parameter
Value
Description
FAX V21 Detect Enable
Yes
FAX CNG Detect Enable
Yes
If CNG detection is not needed, set it to No.
FAX Tone Detect Mode
Caller or Callee
FAX Passthru Method
ReINVITE
FAX Process NSE
No
FAX Passthru Codec
G.711u or G.711a
Depends on the passthrough codec used.
FAX Disable ECAN
No
Echo Canc Enable
Yes
FAX Enable T38
No
Modem Line
No
SG3/V.34 SIP Protocol-Based passthrough FAX Call
Parameter
Value
Description
FAX V21 Detect Enable
Yes
FAX CNG Detect Enable
Yes
If CNG detection is not needed, set it to No.
FAX Tone Detect Mode
Caller or Callee
FAX Passthru Method
ReINVITE
FAX Process NSE
No
FAX Passthru Codec
G.711u or G.711a
Depends on the passthrough codec used.
FAX Disable ECAN
No
Echo Canc Enable
Yes
FAX Enable T38
No
Modem Line
No
SIP Protocol-Based T.38 Relay FAX Call
Parameter
Value
Description
FAX V21 Detect Enable
Yes
FAX CNG Detect Enable
Yes
If CNG detection is not needed, set it to No.
FAX Tone Detect Mode
Caller or Callee
FAX Passthru Method
ReINVITE
FAX Disable ECAN
No
Echo Canc Enable
Yes
FAX Enable T38
No
Modem Line
No
FAX T38 Return to Voice
No
If voice call back is needed after T38 fac call, set it to No.
T38 Relay FAX Failover to SIP Protocol-Based Passthrough FAX call
Parameter
Value
Description
FAX V21 Detect Enable
Yes
FAX CNG Detect Enable
Yes
If CNG detection is not needed, set it to No.
FAX Tone Detect Mode
Caller or Callee
FAX Passthru Method
ReINVITE
FAX Passthru Codec
G.711u or G.711a
Depends on the passthrough codec used.
FAX Disable ECAN
No
If connected to SG3 fax machine, set parameter to "Yes".
Echo Canc Enable
Yes
FAX Enable T38
No
Modem Line
No
FAX T38 Return to Voice
No
G3 Passthrough FAX Call With <FAX Line Toggle Code
Parameter
Value
Description
FAX Disable ECAN
No
Echo Canc Enable
Yes
Modem Line
No
FAX Line Toggle Code
#99
Set it to any preferred code.
NOTE Pre-dial #99 before dialing. Fax call made in this method does not detect fax tone or send out any fax request messages (like NSE or Re-INVITE message) to the remote gateway, which causes interoperability issue with remote side.
Other fax parameters are not relevant to the call.
SG3/V.34 Passthrough FAX Call With <FAX Line Toggle Code
Parameter
Value
Description
FAX Disable ECAN
Yes
Echo Canc Enable
Yes
Modem Line
No
FAX Line Toggle Code
#99
Set it to any preferred code.
NOTE Pre-dial #99 before dialing. Fax call made in this method does not detect fax tone or send out any fax request messages (like NSE or Re-INVITE message) to the remote gateway, which causes interoperability issue with remote side.
Other fax parameters are not relevant to the call.
Passthrough Modem Call With <Modem Line Toggle Code>
Parameter
Value
Description
Modem Line
No
If Modem Line Toggle Code is used to make the modem call, do not set it to Yes
Modem Line Toggle Code
*99
Set it to any preferred code.
NOTE Pre-dial *99 before dialing. Other fax parameters are not relevant to the call.
FAX Disable ECAN
Yes
Passthrough Modem Call With <Modem Line>
Parameter
Value
Description
Modem Line
Yes
FAX Disable ECAN
Yes
NOTE Other fax parameters are not relevant to the call.
Passthrough Modem Call With Manual Configuration
Parameter
Value
Description
Preferred Codec
G.711u or G.711a
Depends on the passthrough codec used.
FAX V21 Detect Enable
No
FAX CNG Detect Enable
No
Echo Canc Enable
No
Silence Supp Enable
No
FAX Enable T38
No
FAX Process NSE
No
Modem Line
No
Jitter Buffer Adjustment
No
Network Jitter Level
High
Dial Plan
The default dial plan script for the line is as follows: (*xx|[3469]11|0|00|[2- 9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.)
Each parameter is separated by a semi-colon (;)
Example 1:
*1xxxxxxxxxx<:@fwdnat.pulver.com:5082;uid=jsmith;pwd=xy z
The syntax for a dial plan expression is described in the table below.
Dial Plan Entry
Functionality
*xx
Allow arbitrary 2 digit star code
[3469]11
Allow x11 sequences
0
Operator
00
Int'l Operator
[2-9]xxxxxx
US local number
1xxx[2-9]xxxxxx
US 1 + 10-digit long distance number
xxxxxxxxxxxx.
Everything else
Gateway Accounts
Field
Description
Gateway1/2/3/4
The first of 4 gateways that can be specified to be used in the <Dial Plan> to facilitate call routing specification (that overrides the given proxy information). This gateway is represented by gw1 in the <Dial Plan>. For example, the rule 1408xxxxxxx<:@gw1> can be added to the dial plan such that when the user dials 1408+7digits, the call will be routed to Gateway 1. Without the <:@gw1> syntax, all calls are routed to the given proxy by default (except IP dialing). Default setting: blank
GW1/2/3/4 NAT Mapping Enable
If enabled, the ATA uses NAT mapping when contacting Gateway 1. Default setting: no
GW1/2/3/4 Auth ID
This value is the authentication user-id to be used by the ATA to authenticate itself to Gateway 1. Default setting: blank
GW1/2/3/4 Password
This value is the password to be used by the ATA to authenticate itself to Gateway 1. Default setting: blank
VoIP Fallback to PSTN section
Field
Description
Auto PSTN Fallback
If enabled, the ATA automatically routes all calls to the PSTN gateway when the Line 1 proxy is down (registration failure or network link down). Default setting: yes
FXS Port Polarity Configuration
Field
Description
Idle Polarity
Polarity before a call is connected: Forward or Reverse. Default setting: Forward
Caller Conn Polarity
Polarity after an outbound call is connected: Forward or Reverse. Default setting: Forward.
Callee Conn Polarity
Polarity after an inbound call is connected: Forward or Reverse. Default setting: Forward
PSTN (LINE Port)
Use the Voice > PSTN page to configure the settings for calls through the LINE (PSTN) port.
To open this page: Click Voice on the menu bar, and then click PSTN in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
Field
Description
PSTN Line Enable
To enable this line for service, select yes. Otherwise, select no. Default setting: yes
Incoming Handset List
The devices that ring when an incoming call is received. Default setting: fxs,1,2,3,4,5,6,7,8,9,10
Network Settings
Field
Description
SIP ToS/DiffServ Value
TOS/DiffServ field value in UDP IP packets carrying a SIP message. Default setting: 0x68
SIP CoS Value
CoS value for SIP messages. Valid values are 1 through 7. Default setting: 3
RTP ToS/DiffServ Value
ToS/DiffServ field value in UDP IP packets carrying RTP data. Default setting: 0xb8
RTP CoS Value
CoS value for RTP data. Valid values are 1 through 7. Default setting: 6
Network Jitter Level
Determines how jitter buffer size is adjusted by the ATA device. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high. Default setting: low
Jitter Buffer Adjustment
Controls how the jitter buffer should be adjusted. Select the appropriate setting: up and down, up only, down only, or disable. Default setting: yes
SIP Settings
Field
Description
SIP Transport
The TCP choice provides "guaranteed delivery," which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. As a result, TCP overcomes the main disadvantages of UDP. In addition, for security reasons, most corporate firewalls block UDP ports. With TCP, new ports do not need to be opened or packets dropped, because TCP is already in use for basic activities such as Internet browsing or e-commerce. Options are: UDP, TCP, TLS. The default is UDP. Default setting: UDP
SIP Port
Port number of the SIP message listening and transmission port. Default setting: 5060
SIP 100REL Enable
To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. Default setting: no
EXT SIP Port
The external SIP port number. Default setting: 5061
SIP 100REL Enable
To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. Default setting: no
EXT SIP Port
The external SIP port number. Default setting: blank
Auth Resync-Reboot
If this feature is enabled, the ATA device authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no. Default setting: yes
SIP Proxy-Require
The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported. Enter the appropriate header in the field provided.
SIP Remote-Party-ID
To use the Remote-Party-ID header instead of the From header, select yes. Otherwise, select no. Default setting: no
SIP GUID
The Global Unique ID is generated for each line for each device. When it is enabled, the ATA device adds a GUID header in the SIP request. The GUID is generated the first time the unit boots up and stays with the unit through rebooting and even factory reset. This feature was requested by Bell Canada (Nortel) to limit the registration of SIP accounts. Default setting: no
SIP Debug Option
SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows:
· none—No logging.
· 1-line—Logs the start-line only for all messages.
· 1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses.
· 1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses.
· 1-line excl. REG—Logs the start-line only for all messages except REGISTER requests/responses.
· 1-line excl. OPT|NTFY|REG—Logs the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses.
· full—Logs all SIP messages in full text.
· full excl. OPT—Logs all SIP messages in full text except OPTIONS requests/responses.
· full excl. NTFY—Logs all SIP messages in full text except NOTIFY requests/responses.
· full excl. REG—Logs all SIP messages in full text except REGISTER requests/responses.
· full excl. OPT|NTFY|REG—Logs all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/responses.
Default setting: none
RTP Log Intvl
The interval for the RTP log. Default setting: 0
Restrict Source IP
If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, 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. If configured, the ATA will drop all packets sent to its SIP Ports originated from an untrusted IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured Proxy (or Outbound Proxy if UseOutbound Proxy is yes). Default setting: no
Referor Bye Delay
Controls when the ATA device 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. Default setting: 4
Refer Target Bye Delay
For the Refer Target Bye Delay, enter the appropriate period of time in seconds. Default setting: 0
Referee Bye Delay
For the Referee Bye Delay, enter the appropriate period of time in seconds. Default setting: 0
Refer-To Target Contact
To contact the refer-to target, select yes. Otherwise, select no. Default setting: 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 setting: no
Auth INVITE
When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy. Default setting: no
Use Anonymous with RPID
When set to yes, use "anonymous" in the SIP message when remote party ID is requested in the SIP message. Default setting: yes
Use Local Addr in FROM
The IP address of the local address enclosed in the FROM of the SIP message. Default setting: no
NAT Settings
Field
Description
NAT Mapping Enable
To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. Default setting: no
NAT Keep Alive Enable
To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. Default setting: no
NAT Keep Alive Msg
Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent. Escape sequence of %xx is also accepted. For example, %0d%0a is unescaped into \r\n (CRLF). Default setting: $NOTIFY
NAT Keep Alive Dest
Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current or outbound proxy. Default setting: $PROXY
Proxy and Registration
Field
Description
Proxy
SIP proxy server for all outbound requests. Default setting: blank
Outbound Proxy
SIP Outbound Proxy Server where all outbound requests are sent as the first hop. Default setting: blank
Use Outbound Proxy
Enable the use of Outbound Proxy. If set to no, the Outbound Proxy parameter and Use OB Proxy in Dialog is ignored. Default setting: no
Use OB Proxy In Dialog
Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the Use Outbound Proxy parameter is no, or if the Outbound Proxy parameter is empty. Default setting: yes
Register
Enable periodic registration with the Proxy. This parameter is ignored if the Proxy parameter is not specified. Default setting: yes
Make Call Without Reg
Allow making outbound calls without successful (dynamic) registration by the unit. If No, dial tone will not play unless registration is successful. Default setting: yes
Register Expires
Allow answering inbound calls without successful (dynamic) registration by the unit. If proxy responded to REGISTER with a smaller Expires value, the ATA will renew registration based on this smaller value instead of the configured value. If registration failed with an Expires too brief error response, the ATA will retry with the value given in the Min-Expires header in the error response. Default setting: 3600
Ans Call Without Reg
Expires value in sec in a REGISTER request. ATA will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 - (231 - 1) sec Default setting: yes
Use DNS SRV
If required by your provider, check this box to use DNS SRV lookup for Proxy and Outbound Proxy. Default setting: no
DNS SRV Auto Prefix
If enabled, the ATA will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name. Default setting: no
Proxy Fallback Intvl
This parameter sets the delay (sec) after which the ATA will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. This parameter is useful only if the primary and backup proxy server list is provided to the ATA via DNS SRV record lookup on the server name. (Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the ATA will not attempt to fall back after a fail over). Default setting: 3600
Proxy Redundancy Method
The ATA makes an internal list of proxies returned in DNS SRV records. In normal mode this list will contain proxies ranked by weight and priority.
If the parameter Based on SRV port is configured, the ATA creates a list in normal mode first, and then inspects the port numbers based on the 1st proxy's port on the list. Default setting: Normal
Subscriber Information
Field
Description
Display Name
Display name for caller ID.
User ID
Extension number for this line.
Password
Password for this line.
Use Auth ID
To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password. Default setting: no
Auth ID
The Authentication ID for SIP authentication.
Audio Configuration
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 codec.
Field
Description
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, G726-32, or G729a. Default setting: G711u
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 setting: yes
G729a Enable
To enable the use of the G729a codec at 8 kbps, select yes. Otherwise, select no. Default setting: no
Silence Supp Enable
To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no. Default setting: no
G726-32 Enable
To enable the use of the G726 codec at 32 kbps, select yes. Otherwise, select no. Default setting: no
Echo Canc Enable
To enable the use of the echo canceller, select yes. Otherwise, select no. Default setting: yes
FAX V21 Detect Enable
To enable detection of V21 fax tones, select yes. Otherwise, select no. Default setting: no
Echo Canc Adapt Enable
To enable the echo canceller to adapt, select yes. Otherwise, select no. Default setting: yes
FAX CNG Detect Enable
To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no. Default setting: no
Echo Supp Enable
To enable the use of the echo suppressor, select yes. Otherwise, select no. Default setting: no
FAX Passthru Codec
Select the codec for fax passthrough, G711u or G711a. Default setting: G711u
DTMF Process INFO
To use the DTMF process info feature, select yes. Otherwise, select no. Default setting: yes
FAX Codec Symmetric
To force the ATA device to use a symmetric codec during fax passthrough, select yes. Otherwise, select no. Default setting: yes
DTMF Process AVT
To use the DTMF process AVT feature, select yes. Otherwise, select no. When set to no, the AVT (RFC2833) payload type is not be included in outbound SDP. Default setting: yes
FAX Passthru Method
Select the fax passthrough method: None, NSE, or ReINVITE. Default setting: 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. Default setting: Auto
FAX Process NSE
To use the fax process NSE feature, select yes. Otherwise, select no. Default setting: yes
Symmetric RTP
Enable symmetric RTP operation. If enabled, the ATA sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) the ATA sends RTP to the destination as indicated in the inbound SDP. Default setting: yes
FAX Disable ECAN
If enabled, this feature automatically disables the echo canceller when a fax tone is detected. To use this feature, select yes. Otherwise, select no. Default setting: no
Choose yes to enable or choose no to disable the VoIP-To-PSTN Gateway functionality. Default setting: yes
VoIP Caller Auth Method
The method to authenticate a VoIP Caller to access the PSTN gateway. Choose from none, PIN, or HTTP Digest. Default setting: none
VoIP PIN Max Retry
The number of times that a VoIP caller can attempt to enter a PIN, if the VoIP Caller Auth Method is set to PIN. Default setting: 3
One Stage Dialing
Choose yes to enable or choose no to disable one-stage dialing. This setting applies if the VoIP Caller Auth Method is none or HTTP Digest, or if caller is in the Access List. Default setting: yes
Line 1 VoIP Caller DP
The index number of the dial plan to use when the VoIP Caller is calling from Line 1 of the same ATA during normal operation (in other words, not due to fallback to PSTN service when Line 1 VoIP service is down). the Authentication is skipped for Line 1 VoIP caller. Default setting: 1
VoIP Caller Default DP
The index number of the dial plan to use when the VoIP Caller is not authenticated. Dial plans are configured in the Dial Plans section. Default setting: 1
Line 1 Fallback DP
The index number of the dial plan to use when the VoIP Caller is calling from Line 1 of the same ATA due to fallback to PSTN service when Line 1 VoIP service is down. Default setting: none
VoIP Caller ID Pattern
A comma-separated list of caller phone number patterns that is used to allow or block access to the PSTN gateway based on the caller ID. If the caller ID does not match a specified pattern, access is rejected, regardless of the authentication method. This comparison is applied before the access list is applied. If this parameter is blank (not specified), all callers are considered for VoIP service.
•Use ? to match any single digit.
•Use * to match any number of digits.
Example: 1408*, 1512???1234
In the above example, the caller ID either must start with 1408 or must be an 11-digit numbering starting with 1512 and ending with 1234.
Default setting: blank
VoIP Access List
A comma-separated list of number patterns that is used to allow or block access to the PSTN gateway based on the source IP address. If the IP address matches a specified pattern, service is allowed without further authentication.
Example: 192.168.*.*, 66.43.12.1??.
In the above example, the source IP address either must begin with 192.168 or must be in the range of 66.43.12.100-199.
Default setting: blank
VoIP Caller 1/2/3/4/5/6/7/8 PIN
A PIN number that a VoIP caller can use to access the PSTN gateway, when the VoIP Caller Auth Method is set to PIN. Default setting: blank
VoIP Caller 1/2/3/4/5/6/7/8 DP
The index number of the dial plan to use upon successful entry of the corresponding VoIP Caller PIN. Dial plans are configured in the Dial Plans section. Default setting: 1
VoIP Users and Passwords (HTTP Authentication)
Field
Description
VoIP User 1/2/3/4/5/6/7/8 Auth ID
A user ID that a VoIP Caller can use for authentication by using the HTTP Digest method (in other words, by embedding an Authorization header in the SIP INVITE message sent to the ATA. If the credentials are missing or incorrect, the ATA will challenge the caller with a 401 response). The VoIP caller whose authentication user-id equals to this ID is referred to VoIP User 1 of this ATA.
NOTE: If the caller specifies an authentication user-id that does not match any of the VoIP User Auth ID's, the INVITE will be rejected with a 403 response. Default setting: blank.
VoIP User 1/2/3/4/5/6/7/8 Password
The password to be used with VoIP User 1. The user assumes the identity of VoIP User 1 must therefore compute the credentials using this password, or the INVITE will be challenged with a 401 response Default setting: blank.
VoIP User 1/2/3/4/5/6/7/8 DP
For up to 8 VoIP users, specify the index of the dial plan to be used after successful authentication. Dial plans are configured in the Dial Plans section. If authentication is disabled, the default dial plan is used for all unknown VoIP users. Default setting: 1.
PSTN-To-VoIP Gateway Setup
Field
Description
PSTN-To-VoIP Gateway Enable
Select yes to enable or select no to disable PSTN-To-VoIP Gateway functionality. Default setting: yes
PSTN Caller Auth Method
The method to authenticate a PSTN Caller to access the VoIP gateway. Choose from none or PIN. Default setting: none
PSTN Ring Thru Line 1
To enable ring through to Line 1 based on caller number patterns, choose yes. Otherwise choose no.
Note: For more information about PSTN Caller number patterns, see PSTN Caller ID Pattern.
Default setting: yes
PSTN PIN Max Retry
The number of times that a PSTN caller can attempt to enter a PIN number, if the authentication method is set to PIN. Default setting: 3
PSTN CID for VoIP CID
Choose yes or no. Default setting: no
PSTN CID Number Prefix
A dialing prefix, if needed, to add to the caller ID number on the PBX to ensure that a callback goes to the correct number. Default setting: blank
PSTN Caller Default DP
The index number of the dial plan that is used when the PSTN Caller Auth Method is set to none. Dial plans are configured in the Dial Plans section. Default settings: 1
Line 1 Signal Hook Flash to PSTN
Specify the operation of the hook flash on the analog phone when a PSTN-to-VoIP call is active. Choose Disabled or Double Hook Flash. Default setting: Disabled
PSTN CID Name Prefix
The prefix to add to the caller ID name that is sent to the PBX. Enter the characters to add to the caller ID name. Default setting: blank
PSTN Caller ID Pattern
A comma-separated list of phone number patterns that is used to allow or block access to the VoIP gateway based on the caller ID. If the caller ID does not match a specified pattern, access is rejected, regardless of the authentication method. This comparison is applied before the access list is applied. If this parameter is blank (not specified), all callers are considered for VoIP service.
•Use ? to match any single digit.
•Use * to match any number of digits.
Example: 1408*, 1512???1234
In the above example, the caller ID either must start with 1408 or must be an 11-digit numbering starting with 1512 and ending with 1234.
Default setting: blank
PSTN Access List
A comma-separated list of number patterns that is used to allow or block access to the VoIP gateway based on the destination IP address. If the destination IP address matches a specified pattern, service is allowed without further authentication.
Example: 192.168.*.*, 66.43.12.1??.
In the above example, the IP address either must begin with 192.168 or must be in the range of 66.43.12.100-199.
The default is blank.
PSTN Caller 1/2/3/4/5/6/7/8 PIN
A PIN number that allows a PSTN caller to access to the VoIP gateway. Calls will be subject to the dial plan specified by the corresponding PSTN Caller DP setting (see below). These settings apply when the PSTN Caller Authentication Method parameter is set to PIN. Default setting: blank
PSTN Caller 1/2/3/4/5/6/7/8 DP
The index number of the dial plan to use upon successful entry of the corresponding PSTN Caller PIN. Dial plans are configured in the Dial Plans section. Default setting: 1
PSTN Timer Values (sec)
Field
Description
VoIP Answer Delay
The number of seconds to wait before auto-answering an inbound VoIP call for the FXO account. The range is 0-255. Default setting: 0
VoIP PIN Digit Timeout
After a VoIP caller is prompted for a PIN or enters a digit, the number of seconds to wait for an entry. The range is 0-255. Default setting: 10
PSTN Answer Delay
After an inbound PSTN call starts ringing, the number of seconds to wait before auto-answering the call. The range is 0-255. Default setting: 16
PSTN PIN Digit Timeout
After a PSTN caller is prompted for a PIN or enters a digit, the number of seconds to wait for an entry. The range is 0-255. Default setting: 10
PSTN-To-VoIP Call Max Dur
The limit on the duration of a PSTN-To-VoIP Gateway Call. Unit is in seconds. 0 means unlimited. The range is 0-2147483647. Default setting: 0
PSTN Ring Thru Delay
After a PSTN call starts ringing, the number of seconds to wait before ring through to Line 1. In order for Line 1 to have the caller ID information, this value must be greater than the time required to complete the PSTN caller ID delivery. The range is 0-255. Default setting: 1
VoIP-To-PSTN Call Max Dur
The limit on the duration of a VoIP-To-PSTN Gateway Call. Unit is in seconds. 0 means unlimited. The range is 0-2147483647. Default setting: 0
PSTN Ring Thru CWT Delay
When a call is active and a new PSTN call starts ringing, the number of seconds to wait before ring through to Line 1 with a Call WaitingTone. Default setting: 3
VoIP DLG Refresh Intvl
The interval between (SIP) Dialog refresh messages sent by the ATA to detect if the VoIP call-leg is still up. If this value is set to 0, the VoIP call-leg status will not be checked by the ATA. The refresh message is a SIP ReINVITE, and the VoIP peer must response with a 2xx response. If the VoIP peer does not reply or the response is not greater than 2xx, the ATA will disconnect both call legs automatically. The range is 0-255. Default setting: 0
PSTN Ring Timeout
After a ring burst, the number of seconds to wait before concluding that PSTN ring has ceased. The range is 0-255. Default setting: 5
PSTN Dialing Delay
After hook, the number of seconds to wait before dialing a PSTN number. The range is 0-255. Default setting: 1
PSTN Dial Digit Len
The on/off time when the Gateway transmits digits through the Line (FXO) port. The syntax is on-time/off-time, expressed in seconds. The permitted range is 0.05 to 3.00 (up to two decimal places only). Default setting: .1/.1
PSTN Hook Flash Len
The length of the hook flash in seconds. Default setting: .25
PSTN Disconnect Detection
Field
Description
Detect CPC
Choose yes to enable or choose no to disable this feature. CPC is a brief removal of tip-and-ring voltage. If enabled, the ATA will disconnect both call legs when this signal is detected during a gateway call. Default setting: yes
Detect Polarity Reversal
Choose yes to enable or choose no to disable this feature. If enabled, the ATA will disconnect both call legs when this signal is detected during a gateway call. If it is a PSTN gateway call, the first polarity reversal is ignored and the second one triggers the disconnection. For VoIP gateway call, the first polarity reversal triggers the disconnection. Default setting: yes
Detect PSTN Long Silence
Choose yes to enable or choose no to disable this feature. If enabled, the ATA will disconnect both call legs when the PSTN side has no voice activity for a duration longer than the length specified in the Long Silence Duration parameter during a gateway call. Default setting: no
Detect VoIP Long Silence
Choose yes to enable or choose no to disable this feature. If enabled, the ATA will disconnect both call legs when the VoIP side has no voice activity for a duration longer than the length specified in the Long Silence Duration parameter during a gateway call. Default setting: no
PSTN Long Silence Duration
This value is minimum length of PSTN silence (or inactivity) in seconds to trigger a gateway call disconnection if Detect Long Silence is enabled. Default setting: 30
VoIP Long Silence Duration
This value is minimum length of VoIP silence (or inactivity) in seconds to trigger a gateway call disconnection if Detect Long Silence is enabled. Default setting: 30
PSTN Silence Threshold
This parameter adjusts the sensitivity of PSTN silence detection. Choose from {very low, low, medium, high, very high}. The higher the setting, the easier to detect silence and hence easier to trigger a disconnection. Default setting: medium
Min CPC Duration
Specify the minimum duration of a low tip-and-ring voltage (below 1V) for the Gateway to recognize it as a CPC signal or PSTN line removal. Default setting: 0.2
Detect Disconnect Tone
Choose yes to enable or choose no to disable this feature. If enabled, the ATA will disconnect both call legs when it detects the disconnect tone from the PSTN side during a gateway call. Disconnect tone is specified in the Disconnect Tone parameter, which depends on the region of the PSTN service. Default setting: yes
Disconnect Tone
This value is the tone script which describes to the ATA the tone to detect as a disconnect tone. The syntax follows a standard Tone Script with some restrictions. Default value is standard US reorder (fast busy) tone, for 4 seconds. Default setting: 480@-30,620@-30;4(.25/.25/1+2)
Restrictions:
•Two frequency components must be given. If single frequency is desired, the same frequency is used for both
•The tone level value is not used. -30 (dBm) should be used for now.
•Only 1 segment set is allowed
•Total duration of the segment set is interpreted as the minimum duration of the tone to trigger detection
•6 segments of on/off time (seconds) can be specified. A 10% margin is used to validated cadence characteristics of the tone.
The country of deployment. This setting applies the relevant regional settings for PSTN calls. Default setting: USA
Tip Ring Voltage Adjustment
Voltage adjustment. The choices are 3.1V, 3.2V, 3.35V, and 3.5V. Default setting: 3.5V.
Ring Frequency Min
The lower limit of the ring frequency used to detect the ring signal. Default setting: 10
SPA To PSTN Gain
dB of digital gain (or attenuation if negative) to be applied to the signal sent from the ATA to the PSTN side. The range is -15 to 12. Default setting: 0
Ring Frequency Max
The higher limit of the ring frequency used to detect the ring signal. Default setting: 100
PSTN To SPA Gain
dB of digital gain (or attenuation if negative) to be applied to the signal sent from the PSTN side to the ATA. The range is -15 to 12. Default setting: 0
Ring Validation Time
The minimum signal duration required by the Gateway for recognition as a ring signal. Default setting: 256 ms
Choose from {13.5-16.5, 19.35-2.65, 40.5-49.5} (Vrms). Default setting: 13.5-16.5 Vrms
Line-In-Use Voltage
The voltage threshold at which the ATA assumes the PSTN is in use by another handset sharing the same line (and will declare PSTN gateway service not available to incoming VoIP callers). Default setting: 30
User 1
Use the Voice > User 1 page to set the user preferences for the calls through the PHONE port1 .
To open this page: Click Voice on the menu bar, and then click User 1 in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
Call Forward Settings
Field
Description
Cfwd All Dest
Forward number for Call Forward All Service. Default setting: blank
Cfwd Busy Dest
Forward number for Call Forward Busy Service. Same as Cfwd All Dest. Default setting: blank
Cfwd No Ans Dest
Forward number for Call Forward No Answer Service. Same as Cfwd All Dest. Default setting: blank
Cfwd No Ans Delay/Forward No Ans Delay
Delay in sec before Call Forward No Answer triggers. Same as Cfwd All Dest. Default setting: 20
Selective Call Forward Settings
Field
Description
Cfwd Sel1-8 Caller
Caller number pattern to trigger Call Forward Selective service. When the caller's phone number matches the entry, the call is forwarded to the corresponding Cfwd Selective Destination (Cfwd Sel1-8 Dest).
•Use ? to match any single digit.
•Use * to match any number of digits.
Example: 1408*, 1512???1234
In the above example, a call is forwarded to the corresponding destination if the caller ID either starts with 1408 or is an 11-digit numbering starting with 1512 and ending with 1234.
Default setting: blank
Cfwd Sel1-8 Dest
The destination for the corresponding Call Forward Selective caller pattern (Cfwd Sel1-8 Caller). Default setting: blank
Cfwd Last Caller
The number of the last caller; this caller is actively forwarded to the Cfwd Last Dest via the Call Forward Last service. For more information, see Vertical Service Activation Codes. Default setting: blank
Cfwd Last Dest
The destination for the Cfwd Last Caller.
Block Last Caller
The number of the last caller; this caller is blocked via the Block Last Caller Service. For more information, see Vertical Service Activation Codes. Default setting: blank
Accept Last Caller
The number of the last caller; this caller is accepted via the Accept Last Caller Service. For more information, see Vertical Service Activation Codes. Default setting: blank
Speed Dial Settings
Field
Description
Speed Dial 2-9
Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9. Default setting: blank
Supplementary Service Settings (User)
Field
Description
CW Setting
Call Waiting on/off for all calls. Default setting: yes
Block CID Setting
Block Caller ID on/off for all calls. Default setting: no
Block ANC Setting
Block Anonymous Calls on or off. Default setting: no
DND Setting
DND on or off. Default setting: no
CID Setting
Caller ID Generation on or off. Default setting: yes
CWCID Setting
Call Waiting Caller ID Generation on or off. Default setting: yes
Dist Ring Setting
Distinctive Ring on or off. Default setting: yes
Secure Call Setting
If yes, all outbound calls are secure calls by default, without requiring the user to dial a star code first. Default setting: no
•If Secure Call Setting is set to yes, all outbound calls are secure. However, a user can disable security for a call by dialing *19 before dialing the target number.
•If Secure Call Setting is set to No, the user can make a secure outbound call by dialing *18 before dialing the target number.
•A user cannot force inbound calls to be secure or not secure; that depends on whether the caller has security enabled or not.
Note: This setting is applicable only if Secure Call Serv is set to yes on the line interface. See Line 1 Settings (PHONE Port).
Message Waiting
Setting this value to yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and will survive after reboot or power cycle. Default setting: no
Accept Media Loopback Request
Controls how to handle incoming requests for loopback operation. Default setting: automatic
•never: Never accepts loopback calls; replies 486 to the caller.
•automatic: Automatically accepts the call without ringing.
•manual: Rings the phone first, and the call must be picked up manually before loopback starts. Default setting: Automatic
Media Loopback Mode
The loopback mode to assume locally when making call to request media loopback. Choices are: Source and Mirror. Default setting: source
NOTE If the ATA answers the call, the mode is determined by the caller.
Media Loopback Type
The loopback type to use when making call to request media loopback operation. Choices are Media and Packet. Default setting: media
Note that if the ATA answers the call, then the loopback type is determined by the caller (the ATA always picks the first loopback type in the offer if it contains multiple type)
Distinctive Ring Settings
Field
Description
Ring1 - 8 Caller
Caller number pattern to play Distinctive Ring/CWT 1, 2, 3, 4, 5, 6, 7, or 8. Caller number patterns are matched from Ring 1 to Ring 8. The first match (not the closest match) will be used for alerting the subscriber. The distinctive rings are set on the Regional page. Default setting: blank
Ring Settings
Field
Description
Default Ring
Default ringing pattern, 1-8, for all callers. Default setting: 1
Default CWT
Default CWT pattern, 1-8, for all callers. Default setting: 1
Hold Reminder Ring
Ring pattern for reminder of a holding call when the phone is on-hook. Default setting: 8
Call Back Ring
Ring pattern for call back notification. Default setting: 7
Cfwd Ring Splash Len
Duration of ring splash when a call is forwarded (0 - 10.0s) Default setting: 0
Cblk Ring Splash Len
Duration of ring splash when a call is blocked (0 - 10.0s) Default setting: 0
VMWI Ring Policy
The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the ATA indicating the status of the subscriber's mail box. Three settings are available. Default setting: New VM Available
•New VM Available: Ring as long as there new voicemail messages.
•New VM Becomes Available: Ring at the point when the first new voicemail message is received.
•New VM Arrives: Ring when the number of new voicemail messages increases.
VMWI Ring Splash Len
Duration of ring splash when new messages arrive before the VMWI signal is applied (0 - 10.0s) Default setting: 0
Ring On No New VM
If enabled, the ATA plays a ring splash when the voicemail server sends SIP NOTIFY message to the ATA indicating that there are no more unread voice mails. Some equipment requires a short ring to precede the FSK signal to turn off VMWI lamp. Default setting: no
PSTN User
Use the Voice > PSTN User page to set the user preferences for calls through the LINE port.
To open this page: Click Voice on the menu bar, and then click PSTN in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
PSTN-To-VoIP Selective Call Forward Settings
Field
Description
Cfwd Sel1-8 Caller
Eight PSTN Caller Number Patterns to be blocked for VoIP gateway services or forwarded to a certain VoIP number. If the caller is blocked, the ATA will not auto-answers the call.
Cfwd Sel1-8 Dest
Eight VoIP destinations to forward a PSTN caller matching the Cfwd Sel x Caller parameter. If this entry is blank, the PSTN caller is blocked for VoIP service.
Cfwd Last Caller
The Caller number that is actively forwarded to Cfwd Last Dest by using the Call Forward Last activation code. Default setting: blank
Cfwd Last Dest
Forward number for the Cfwd Last Caller parameter. Default setting: blank
Block Last Caller
ID of caller blocked via the Block Last Caller service. Default setting: blank
Accept Last Caller
ID of caller accepted via the Accept Last Caller service. Default setting: blank
PSTN-To-VoIP Speed Dial Settings
Field
Description
Speed Dial 2-9
The VoIP number to call when the PSTN caller dials the specified digit. Default setting: blank
PSTN Ring Thru Line 1 Distinctive Ring Settings
Field
Description
Ring1-8 Caller
Eight PSTN Caller Number Patterns such that the corresponding ring will be used to ring through Line 1 if the PSTN caller matches this pattern. The ring patterns are configured on the Voice > Regional page. For more information, see Distinctive Ring Patterns. Default setting: blank
PSTN Ring Thru Line 1 Ring Settings
Field
Description
Default Ring
The default ring to be used to ring through Line 1. Choose from {1,2,3,4,5,6,7,8,Follow Line Cfg}. If Follow Line Cfg is selected, the ring is determined by the distinctive ring settings for Line 1. The ring patterns are configured on the Voice > Regional page. For more information, see Distinctive Ring Patterns. Default setting: 2
DECT Line 1 - DECT Line 10
Use the Voice > DECT Line 1~DECT Line 10 pages to configure the settings for calls using Cisco SPA302D handsets.
To open this page: Click Voice on the menu bar, and then click DECT Line 1~10 in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
NOTE When a DECT line receives an incoming from a RU and if no handset is registered to this line or all handsets attached to this line are turned off, the base responds to the RU in 9 seconds with "486 Busy Here."
General
Field
Description
Line Enable
To enable this line for service, select yes. Otherwise, select no. Default setting: yes
Streaming Audio Server (SAS)
Field
Description
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 setting: 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 ATA ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled) Default setting: 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 value is used by the 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 send-only 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 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 will be 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 setting: blank
NAT Settings
Field
Description
NAT Mapping Enable
To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no. Default setting: no
NAT Keep Alive Enable
To send the configured NAT keep alive message periodically, select yes. Otherwise, select no. Default setting: no
NAT Keep Alive Msg
Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent. Escape sequence of %xx is also accepted. For example, %0d%0a is unescaped into \r\n (CRLF). Default setting: $NOTIFY
NAT Keep Alive Dest
Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current or outbound proxy. Default setting: $PROXY
Network Settings
Field
Description
SIP ToS/DiffServ Value
TOS/DiffServ field value in UDP IP packets carrying a SIP message. Default setting: 0x68
SIP CoS Value
CoS value for SIP messages. Valid values are 0 through 7. Default setting: 3
RTP ToS/DiffServ Value
ToS/DiffServ field value in UDP IP packets carrying RTP data. Default setting: 0xb8
RTP CoS Value [0- 7]
CoS value for RTP data. Valid values are 0 through 7. Default setting: 6
Network Jitter Level
Determines how jitter buffer size is adjusted by the ATA. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high. Default setting: high
Jitter Buffer Adjustment
Choose yes to enable or no to disable this feature. Default setting: yes
SIP Settings
Field
Description
SIP Transport
The TCP choice provides "guaranteed delivery", which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. As a result, TCP overcomes the main disadvantages of UDP. In addition, for security reasons, most corporate firewalls block UDP ports. With TCP, new ports do not need to be opened or packets dropped, because TCP is already in use for basic activities such as Internet browsing or e-commerce. Options are: UDP, TCP, TLS. Default setting: UDP
SIP Port
Port number of the SIP message listening and transmission port. Default setting: 5060
SIP 100REL Enable
To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. Default setting: no
EXT SIP Port
The external SIP port number. Default setting: blank
Auth Resync-Reboot
If this feature is enabled, the ATA authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no. Default setting: yes
SIP Proxy-Require
The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported. Enter the appropriate header in the field provided. Default setting: blank
SIP Remote-Party-ID
To use the Remote-Party-ID header instead of the From header, select yes. Otherwise, select no. Default setting: yes
SIP GUID
This feature limits the registration of SIP accounts. The Global Unique ID is generated for each line for each ATA. When it is enabled, the ATA adds a GUID header in the SIP request. The GUID is generated the first time the unit boots up and stays with the unit through rebooting and even factory reset. Default setting: no
SIP Debug Option
SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. The choices are described below. Default setting: none
•none—No logging.
•1-line—Logs the start-line only for all messages.
•1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses.
•1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses.
•1-line excl. REG—Logs the start-line only for all messages except REGISTER requests/responses.
•1-line excl. OPT|NTFY|REG—Logs the start-line only for all messages except OPTIONS, NOTIFY, and REGISTER requests/responses.
•full—Logs all SIP messages in full text.
•full excl. OPT—Logs all SIP messages in full text except OPTIONS requests/responses.
•full excl. NTFY—Logs all SIP messages in full text except NOTIFY requests/responses.
•full excl. REG—Logs all SIP messages in full text except REGISTER requests/responses.
•full excl. OPT|NTFY|REG—Logs all SIP messages in full text except for OPTIONS, NOTIFY, and REGISTER requests/responses.
RTP Log Intvl
The interval for the RTP log. Default setting: 0
Restrict Source IP
If Lines 1 and 2 use the same SIP Port value and the Restrict Source IP feature is enabled, 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. If configured, the ATA will drop all packets sent to its SIP Ports originated from an untrusted IP address. A source IP address is untrusted if it does not match any of the IP addresses resolved from the configured Proxy (or Outbound Proxy if Use Outbound Proxy is yes) Default setting: no
Referor Bye Delay
Controls when the ATA 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. Default setting: 4
Refer Target Bye Delay
For the Refer Target Bye Delay, enter the appropriate period of time in seconds. Default setting: 0
Referee Bye Delay
For the Referee Bye Delay, enter the appropriate period of time in seconds. Default setting: 0
Refer-To Target Contact
To contact the refer-to target, select yes. Otherwise, select no. Default setting: no
Sticky 183
If this feature is enabled, the ATA 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 setting: no
Use Anonymous With RPID
When set to yes, use "anonymous" in the SIP message. Default setting: yes
Use Local Addr In From
Use the local ATA IP address in the SIP FROM message. Default setting: no
Auth INVITE
When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy. Default setting: no
Reply 182 On Call Waiting
When enabled, the ATA replies with a SIP182 response to the caller if it is already in a call and the line is off-hook. To use this feature select yes. Default setting: no
Call Feature Settings
Field
Description
Blind Attn-Xfer Enable
Enables the ATA 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 ATA 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 setting: 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. Default setting: blank
Xfer When Hangup Conf
Makes the ATA perform a transfer when a conference call has ended. Select yes or no from the drop-down menu. Default setting: 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. For example: conf@mysefver.com:12345 or conf (which uses the Proxy value as the domain). Default setting: blank
Conference Bridge Ports
Select the maximum number of conference call participants. The range is 3 to 10. Default setting: 3
Voice Mail Number
The phone number for the voice mail system. Default setting: blank
Mailbox ID
Enter the ID number of the mailbox for this line. Default setting: blank
Proxy and Registration
Field
Description
Proxy
SIP proxy server for all outbound requests. Default setting: blank
Outbound Proxy
SIP Outbound Proxy Server where all outbound requests are sent as the first hop. Default setting: blank
Use Outbound Proxy
Enable the use of Outbound Proxy. If set to no, the Outbound Proxy parameter and Use OB Proxy in Dialog is ignored. Default setting: no
Use OB Proxy In Dialog
Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the Use Outbound Proxy parameter is no, or if the Outbound Proxy parameter is empty. Default setting: yes
Register
Enable periodic registration with the Proxy. This parameter is ignored if the Proxy parameter is not specified. Default setting: yes
Make Call Without Reg
Allow making outbound calls without successful (dynamic) registration by the unit. If No, dial tone will not play unless registration is successful. Default setting: yes
Register Expires
Allow answering inbound calls without successful (dynamic) registration by the unit. If proxy responded to REGISTER with a smaller Expires value, the ATA will renew registration based on this smaller value instead of the configured value. If registration failed with an Expires too brief error response, the ATA will retry with the value given in the Min-Expires header in the error response. Default setting: 3600
Ans Call Without Reg
Expires value in sec in a REGISTER request. ATA will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 - (231 - 1) sec Default setting: yes
Use DNS SRV
If required by your provider, check this box to use DNS SRV lookup for Proxy and Outbound Proxy. Default setting: no
DNS SRV Auto Prefix
If enabled, the ATA will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name. Default setting: no
Proxy Fallback Intvl
This parameter sets the delay (sec) after which the ATA will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. This parameter is useful only if the primary and backup proxy server list is provided to the ATA via DNS SRV record lookup on the server name. (Using multiple DNS A record per server name does not allow the notion of priority and so all hosts will be considered at the same priority and the ATA will not attempt to fall back after a fail over). Default setting: 3600
Proxy Redundancy Method
The ATA makes an internal list of proxies returned in DNS SRV records. In normal mode this list will contain proxies ranked by weight and priority.
If the parameter Based on SRV port is configured, the ATA creates a list in normal mode first, and then inspects the port numbers based on the 1st proxy's port on the list. Default setting: Normal
Voice Mail Server
The URL or IP address of the voice mail server.
Mailbox Subscribe Expires
The subscription interval for voicemail message waiting indication. When this time period expires, the ATA sends another subscribe message to the voice mail server. Default: 2147483647
Subscriber Information
Field
Description
Display Name
Display name for caller ID.
User ID
Extension number for this line.
Password
Password for this line.
Use Auth ID
To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password. Default setting: no
Auth ID
The Authentication ID for SIP authentication.
Directory Number
The number for this line.
Resident Online Number
This setting allows you to associate a "local" telephone number with this line using a valid Skype Online Number from Skype. Calls made to that number will ring your phone. Enter the number without spaces or special characters. Default setting: blank
Supplementary Service Subscription
The ATA provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service. A supplementary service should be disabled if a) the user has not subscribed for it, or b) the Service Provider intends to support similar service using other means than relying on the ATA.
Field
Description
Call Waiting Serv
Enable Call Waiting Service. Default setting: yes
Block CID Serv
Enable Block Caller ID Service. Default setting: yes
Block ANC Serv
Enable Block Anonymous Calls Service Default setting: yes
Dist Ring Serv
Enable Distinctive Ringing Service Default setting: yes
Cfwd All Serv
Enable Call Forward All Service Default setting: yes
Cfwd Busy Serv
Enable Call Forward Busy Service Default setting: yes
Cfwd No Ans Serv
Enable Call Forward No Answer Service Default setting: yes
Enable Forward Last Call Service Default setting: yes
Block Last Serv
Enable Block Last Call Service Default setting: yes
Accept Last Serv
Enable Accept Last Call Service Default setting: yes
DND Serv
Enable Do Not Disturb Service Default setting: yes
CID-Serv
Enable Caller ID Service Default setting: yes
CWCID Serv
Enable Call Waiting Caller ID Service Default setting: yes
Call Return Serv
Enable Call Return Service Default setting: yes
Call Redial Serv
Enable Call Redial Service.
Call Back Serv
Enable Call Back Service.
Three Way Call Serv
Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer. Default setting: yes
Three Way Conf Serv
Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer. Default setting: yes
Attn Transfer Serv
Enable Attended Call Transfer Service. Three Way Conference is required for Attended Transfer. Default setting: yes
Unattn Transfer Serv
Enable Unattended (Blind) Call Transfer Service. Default setting: yes
MWI Serv
Enable MWI Service. MWI is available only if a Voice Mail Service is set-up in the deployment. Default setting: yes
VMWI Serv
Enable VMWI Service (FSK) Default setting: yes
Speed Dial Serv
Enable Speed Dial Service. Default setting: yes
Secure Call Serv
Secure Call Service. If this feature is enabled, a user can make a secure call by entering an activation code (*18 by default) before dialing the target number. Then audio traffic in both directions is encrypted for the duration of the call. Default setting: yes
For more information about star code settings, see Vertical Service Activation Codes. To enable secure calling by default, without requiring a star code, set the user's Secure Call Setting to yes. See User 1.
Referral Serv
Enable Referral Service. See the Referral Services Codes parameter For more information. Default setting: yes
Feature Dial Serv
Enable Feature Dial Service. See the Feature Dial Services Codes parameter For more information. Default setting: yes
Service Announcement Serv
Enable Service Announcement Service. Default setting: no
Audio Configuration
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 codec.
Field
Description
Preferred Codec, Second Preferred Codec, Third Preferred Codec
Up to three codecs to be used for all calls from this handset, listed order of preference. The actual codec used in a call still depends on the outcome of the codec negotiation protocol. Select one of the following: G711u, G711a, G726-32, G729a, or G722. Default setting for Preferred Codec: G711u Default setting for Second and Third Preferred Codec: Unspecified
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 setting: yes
Use Remote Pref Codec
To use the preferred codec specified by the remote peer, select yes. Otherwise, select no. Default setting:
Codec Negotiation
Specify the codecs for codec negotiation: Default or List All. Default setting: Default
Silence Supp Enable
To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no. Default setting: no
Silence Threshold
Select the appropriate setting for the threshold: high, medium, or low. Default setting: medium
G729a Enable
To enable the use of the G729a codec at 8 kbps, select yes. Otherwise, select no. Default setting: no
Echo Canc Enable
To enable the use of the echo canceller, select yes. Otherwise, select no. Default setting: yes
G726-32 Enable
To enable the use of the G726 codec at 32 kbps, select yes. Otherwise, select no. Default setting: no
G722 Enable
To enable the use of the G722 codec at 32 kbps, select yes. Otherwise, select no. Default setting: no
DTMF Process INFO
To use the DTMF process info feature, select yes. Otherwise, select no. Default setting: yes
DTMF Process AVT
To use the DTMF process AVT feature, select yes. Otherwise, select no. When set to no, the AVT (RFC2833) payload type is not be included in outbound SDP. Default setting: yes
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. Default setting: Auto
DTMF Tx Mode
DTMF Detection Tx Mode is available for SIP information and AVT. Options are: Strict or Normal. Default setting: Strict for which the following are true:
•A DTMF digit requires an extra hold time after detection.
•The DTMF level threshold is raised to -20 dBm.
The minimum and maximum duration thresholds are:
•strict mode for AVT: 70 ms
•normal mode for AVT: 40 ms
•strict mode for SIP info: 90 ms
•normal mode for SIP info: 50 ms
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. Default setting: None
Symmetric RTP
Enable symmetric RTP operation. If enabled, the ATA sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) the ATA sends RTP to the destination as indicated in the inbound SDP. Default setting: yes
Dial Plan
Field
Description
Dial Plan
The allowed number patterns for outbound calls. For information about the dial plan syntax, see Configuring Dial Plans. Default setting: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
PSTN Fallback Dial Plan
The dial plan used when PSTN fallback is enabled and in use. Default setting: (S0<:@gw0>)
Enable IP Dialing
Enable or disable IP dialing. If IP dialing is enabled, one can dial [userid@] a.b.c.d[:port], where `@', `.', and `:' are dialed by entering *, user-id must be numeric (like a phone number) and a, b, c, d must be between 0 and 255, and port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are optional. If the user-id portion matches a pattern in the dial plan, then it is interpreted as a regular phone number according to the dial plan. The INVITE message, however, is still sent to the outbound proxy if it is enabled. Default setting: no
Emergency Number
Comma separated list of emergency number patterns. If outbound call matches one of the pattern, the ATA will disable hook flash event handling. The condition is restored to normal after the call ends. Blank signifies that there is no emergency number. Maximum number length is 63 characters. Default setting: blank
Incoming Handset List
The devices that ring when an incoming call is received. Default setting: fxs,1,2,3,4,5,6,7,8,9,10
Call Forward Settings
Field
Description
Cfwd All Dest
Forward number for Call Forward All Service. Default setting: blank
Cfwd Busy Dest
Forward number for Call Forward Busy Service. Same as Cfwd All Dest. Default setting: blank
Cfwd No Ans Dest
Forward number for Call Forward No Answer Service. Same as Cfwd All Dest. Default setting: blank
Cfwd No Ans Delay
Delay in sec before Call Forward No Answer triggers. Same as Cfwd All Dest. Default setting: 20
Supplementary Service Settings
Field
Description
Secure Call Setting
If yes, all outbound calls are secure calls by default, without requiring the user to dial a star code first. Default setting: no
•If Secure Call Setting is set to yes, all outbound calls are secure. However, a user can disable security for a call by dialing *19 before dialing the target number.
•If Secure Call Setting is set to No, the user can make a secure outbound call by dialing *18 before dialing the target number.
•A user cannot force inbound calls to be secure or not secure; that depends on whether the caller has security enabled or not.
Note: This setting is applicable only if Secure Call Serv is set to yes on the line interface. See Line 1 Settings (PHONE Port).
Message Waiting
Setting this value to yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and will survive after reboot or power cycle. Default setting: no
Accept Media Loopback Request
Controls how to handle incoming requests for loopback operation. Default setting: automatic
•never: Never accepts loopback calls; replies 486 to the caller.
•automatic: Automatically accepts the call without ringing.
•manual: Rings the phone first, and the call must be picked up manually before loopback starts. Default setting: Automatic
Media Loopback Mode
The loopback mode to assume locally when making call to request media loopback. Choices are: Source and Mirror. Default setting: source
NOTE If the ATA answers the call, the mode is determined by the caller.
Media Loopback Type
The loopback type to use when making call to request media loopback operation. Choices are Media and Packet. Default setting: media
Note that if the ATA answers the call, then the loopback type is determined by the caller (the ATA always picks the first loopback type in the offer if it contains multiple type)
DECT User
Use the Voice > DECT User page to set the user preferences for calls using Cisco SPA302D handsets.
To open this page: Click Voice on the menu bar, and then click DECT User in the navigation tree. Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.
General
Field
Description
Call Park Enable
Enables or disables Call Park. Default setting: No
Call Pickup Enable
Enables or disables Call Pickup. Default setting: No
Call Group Pickup Enable
Enables or disables Group Pickup. Default setting: No
Handset 1
Field
Description
Handset Name
Name of the handset can be configured either from the SPA232D web GUI or from the handset GUI. The name can support only alphabets and numbers and the maximum length is 10.
Default setting: Handset 1.
Similary for Handset 2, 3, 4, and 5, the default setting is Handset 2, 3, 4, 5, respectively.
NOTE When multiple handset names are to be changed, they are sent to CSS at a time with a format like this:
5#1: Name1#2: Name2#3: Name3#4: Name4#5: Name5
For example: If 5 handsets are requesting name provisioning simultaneously, and their names are "Hank", "White", "Pinkman", "Gus" and "Skyler" from handset1 to handset5 respectively. Then the format as following:
5#1:Hank#2:White#3:Pinkman#4:Gus#5:Skyler
If the handset name is changed from handset in Settings>Handset Settings>Handset Name, it will be sent back to base and updated on the web page.
Outgoing DECT Lines
A comma-separated list of the index numbers (1~10) for the lines that are available from this handset for an outgoing call. These lines will be listed on the phone screen when the user displays the call options or holds down the green call button.
Example: 1,2,8 In this example, a user can select DECT line 1, 2, or 8 for an outbound call.
Default setting: 1
Note: You also can choose these lines from theDECT Handset Outgoing Line Selection section of the Quick Setup page.
Failover
When this feature is enabled and a call fails through the selected line, the ATA automatically attempts to place the call over another enabled DECT/PSTN line. Select yes to enable this feature or select no to disable it.
Default setting: no
NOTE Cisco SPA232D now supports PSTN to DECT and DECT to PSTN outgoing line failover.
Deregister
To deregister a handset, select yes. After you submit the settings and the voice module reboots, then the handset is deregistered. At that point, this parameter is reset to the default value. Default setting: no
Bound IPEI
Enter the device's IPEI number (a unique hardware identifier comparable to a MAC address) if you want to bind this device to the specified handset ID, such as Handset 3. The IPEI can be found in the Settings > Phone Info menu on the handset. Default setting: blank
Handset Auto Update
To enable handset auto update select yes, otherwise select no.
Default setting: yes
NOTE When enabled, one of the following events triggers software update notification from SPA232D to handset:
· SPA232D reboots
· New handset subscribes to SPA232D
· Registered handset comes in range
a. Five minutes after the firmware update notification from SPA232D, the handset gets the version number from SPA232D when it is idle.
b. If the version retrieved from SPA232D is different from its local version number, the HS displays a message Software Update in 30 Seconds with 2 options—Update and Delay.
· If Delay is selected, the firmware update window pops up again in 1 hour.
· If Update is selected, the handset performs the upgrade immediately.
c. If no soft-key is selected, the handset upgrades automatically in 30 seconds.
If the upgrade fails for some reason, a splash window displays "System busy. Will retry in 5 minutes."
Feedback
SIP
