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Use status station to see the internal software state information for a specific station. Use the information for diagnosis and to help locate the facilities to which the station is communicating.
status station x
Extension number per dial plan.
Field |
Description |
|---|---|
Active Coverage Option |
Specifies the active coverage path. |
Active call(s) |
Displays the current status of the agent on an active call.
|
Administered Type |
Administered station type. |
Busied-out |
Displays the service state of the station.
|
Call Parked |
Whether the station has a call parked (y/n) |
CTI Monitoring Active |
Displays the connection status of the CTI controller.
|
Connected Ports |
Port locations of the facilities to which the telephone/softphone is connected: cabinet-carrier-slot-circuit
Note:
If a station is connected to a QSIG over SIP trunk, you can view the involved port of the SIP trunk. However, you cannot view the port of the QSIG trunk since the port is not involved in the media connection. |
Connected Type |
The type of telephone connected to this port. |
Download Status |
Status of soft key download. |
EC500 Status |
enabled — Extension to Cellular is enabled disabled — Extension to Cellular is disabled |
Extension |
Station or attendant extension. |
Group Cntrl Restr |
One or two of the following:
|
Limit Incoming Calls |
State of the Limit Number of Concurrent Call (LNNC). |
Message Waiting |
Whether there is a message waiting for the station.
|
Network Region |
The network region (NR) that determines whether Communication Manager triggers the IGAR or DPT feature:
|
Parameter Downlaod |
Current status of downloading terminal parameter information.
|
Port |
Port location of the endpoint. |
Reachability Polling Active |
Communication Manager will poll the station with SIP OPTIONS messages directly by the Reachability feature in either of the following cases and the Reachability Polling Active field on status station shall show yes:
|
Reg Subscription Active |
Displays the registration subscription state of the station on Session Manager. This field is active only when CTI is monitoring the station and SIP Reachability is disabled for the station or system-wide.
|
Ring Cut Off Act |
Whether ring cut-off is activated (y/n). |
SAC Activated |
Whether Send All Calls is activated on this extension (y/n). |
Service Link Extension |
Extension for the softphone off-premise destination on a telecommuter configuration. |
Service Link Port |
Shows the port used to establish a connection to the softphone off-premise destination on a telecommuter configuration. |
Service State |
State of the telephone endpoint:
For SIP stations, the following fields are available:
|
Shd TCP Signaling Status |
Indicates whether a TCP signaling channel is established for the endpoint on a shared control connection. Displayed only for IP Telephones and Softphones.
|
Softphone Port |
The port for the softphone controlling an IP telephone. |
TCP Signal Status |
Indicates whether a TCP signaling channel is established for the endpoint. Displayed only for IP Telephones and Softphones.
|
User Cntrl Restr |
One or two of the following
|
Field |
Description |
|---|---|
General Status |
|
Connected Station Information |
|
Part ID Number |
Part ID Number (comcode) of the telephone. If this field shows Unavailable, the software was unable to determine the Part ID Number. |
Serial Number |
Serial number of the telephone.
|
Station Lock Active |
Indicates whether the station has been locked using Station Lock. |
TOD Station Lock |
Indicates whether the station is within a Time of Day lock interval. |
Off-PBX Service State |
|
Hospitality Status |
|
Awaken At |
Time that Automatic Wakeup Call is scheduled. The time is based on the location of the station. |
User DND |
activated/not activated Status of Do Not Disturb |
Group DND |
activated/not activated Status of Do Not Disturb |
Room Status |
Whether a room is occupied or not
|
Languages |
Communication Manager displays the language that you have downloaded on a 9404 or 9408 telephones. The default value of this field is blank. |
Language File Version |
Communication Manager displays the version of the language that you have downloaded on a 9404 or 9408 telephones. The default value of this field is blank. |
CF Destination Ext |
Call forwarding destination extension, if any.
|
Enhanced Call Forwarding Destinations |
Up to six Enhanced Call Forwarding destination numbers for Unconditional, Busy, or No Rely ECF types, and the internal or external destination for each. |
Team Button Monitoring Stations |
Shows all monitoring extensions that supervise the monitored station |
Field |
Description |
|---|---|
Native Name Scripts |
N/A, or on a Unicode-enabled station, the script used for the native name of the station: 0x plus the hex value of the script tag, and up to 7 language acronyms. |
Display Message Scripts |
N/A, or on a Unicode-enabled station, the script used for the display language of the station: 0x plus the hex value of the script tag, and up to 7 language acronyms. |
Station Supported Scripts |
N/A, or on a Unicode-enabled station, the script supported by the telephone: 0x plus the hex value of the script tag based on the scripts supported by Unicode capable endpoints, and up to 7 language acronyms. |
Field descriptions are the same as status station, pages 1 and 2 field descriptions.
Field |
Description |
|---|---|
Call Control Signaling — Call Control Signaling fields display the information for both endpoints when an IP telephone and IP soft phone are in service on the same extension simultaneously (shared-control). |
|
Switch-End IP Signaling Loc |
The procor serving the switch end of the IP signaling link. The port number is always 17 for the ethernet port of the procor. This field is blank for an unregistered H.323 station. |
H.245 Port |
The procor port serving the H.245 signaling link. The port number is 17 for the ethernet port. This field is blank for an unregistered H.323 station, a VPhone, or when H.245 tunneling in Q.931 is used. |
Switch-end: IP Address and Port |
Switch-end Q.931 (H.323 station) or CCMS (IP SoftPhone) IP signaling address and IP port. IP Port has a decimal value 0-65535. This field is blank for an unregistered H.323 station. |
V4 Set-End: IP address and Port |
Set-end IPv4 IP signaling address and IP port. IP port is a decimal value 0-65535. |
V6 Set-End: IP address and Port |
Set-end IPv6 IP signaling address and IP port. IP port is a decimal value 0-65535. |
Node Name (for Call Control) |
Label administered for an IP address. |
Region (for Call Control) |
A number given to a set of IP addresses to indicate they have a common set of characteristics. |
H.245 Near |
Switch-end H.245 IP signaling address and TCP/IP port. IP Port is 0-65535. This field is blank for IP SoftPhone endpoints, endpoints using Q.931 tunneling of H.245, and unregistered or inactive H.323 stations. |
H.245 Set |
Set-end H.245 IP signaling address and IP Port. IP Port is 0-65535. This field is blank for IP SoftPhone endpoints, endpoints using Q.931 tunneling of H.245, and unregistered or inactive H.323 stations. |
Audio Channel fields display the information for only the IP Telephone when an IP telephone and IP soft phone are in service on the same extension simultaneously. Product Information fields show the information for both endpoints for soft phone and Telephone Shared Control.
Field |
Description |
|---|---|
xxxxxxx Audio |
The type of audio active for the station. xxxxxxxx is one of these values: G.711-MU, G711-A, G.729A, G.723.1-6.3, G.723.1-5.3, PCM, 711-MU, 711-A, 729, 729A, 729B, 729-AB, or 723.1-5.3/6.3.
|
Port/Shared Port |
The physical port used to provide audio path for the endpoint. The port represents a media gateway or media server port for H.323 stations, or for stations using a H.323 endpoint for audio. An idle IP SoftPhone or H.323 station with an as-needed service link shows no audio port. Blank — stations with no established audio path, or for telecommuter stations. |
Other-end IP Addr and Port |
Other-end IP audio address and IP port. Port is a decimal value 0–65535. This field is blank for an idle station with no audio link, or for a telecommuter IP SoftPhone with circuit-switched audio link. |
Set-end IP Addr and Port |
Set-end IP audio address and IP port. Port is a decimal value 0–65535. This field is blank for an idle station with no audio link, or for a telecommuter IP SoftPhone with circuit-switched audio link. |
Node Name |
Label administered for an IP address. |
Network Region |
A number given to a set of IP addresses to indicate they have a common set of characteristics. |
Audio Connection Type |
Audio codec selected.
|
MAC Address |
The Media Access Control (MAC) address received from the telephone when the telephone registers. This field is displayed in the second column of the screen when the Shared Port field is populated.
|
Field |
Description |
|---|---|
Product ID and Release |
Identifier submitted by the endpoint during registration, and the release number of the endpoint that is provided to the gatekeeper upon registration. This field is displayed only for Avaya or Lucent products. This field is displayed in the second column of the screen when the Shared Port field is populated. Value is one of the 30 enabled product IDs administered on the system-parameters customer-options screen, including:
|
H.245 Tunneled in Q.931 |
y — H.245 is contained within Q.931, and H.245 fields are not displayed. Does not apply to IP SoftPhone endpoints. This field is displayed in the second column of the screen when the Shared Port field is populated. |
Registration Status |
Identifies the registration and authentication status of the IP endpoint. An unregistered IP SoftPhone cannot be identified as an IP endpoint. This field is displayed in the second column of the screen when the Shared Port field is populated.
|
MAC Address |
The Media Access Control (MAC) address received from the telephone when the telephone registers. This field is displayed in the second column of the screen when the Shared Port field is populated.
|
Authentication Type |
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers. When you register the stations with security profile H323TLS, status station shows Authentication Type as TLS. |
Native NAT Address |
Specifies the network address translation (NAT) IP address of the endpoint when a network device provides the network address translation function for the endpoint. The network device provides the NAT address of the endpoint at the time of registration. The field is blank if the NAT address is not known. This field is displayed in the second column of the screen when the Shared Port field is populated. |
ALG-NAT WAN IP Address |
This field is populated only when a special application turned on. |
Media Encryption |
aes indicates Advanced Encryption Standard encryption, the standard used by U.S. government to protect sensitive (unclassified) information. Reduces circuit-switched to IP call capacity by 25%. aea indicates Avaya Encryption Algorithm. Not as secure as AES. none indicates an unencrypted media stream. |
Field |
Description |
|---|---|
Main |
Main is the phone which did an ordinary H.323 registration. This phone did not register in shared control mode. |
Shared Control Ports |
Shared Control ports register more than one H.323 endpoint to the same Communication Manager extension number. Usually the endpoints are a desk phone and a softphone, but they can include an AES application. |
Node Name |
Node name is a label administered for an IP address. |
Region |
Region is a bunch of IP destinations that have a lot in common with each other. |
Field |
Description |
|---|---|
Grp |
Hunt Group number |
Mod |
ACD Work Mode |
On ACD Call |
Whether the agent is on an ACD call (y/n) |
Occupancy |
Occupancy of the agent logged in at the specified station |
Network Status fields show information for only the IP Telephone when an IP telephone and IP Softphone are in service on the same extension simultaneously.
Field |
Description |
|---|---|
Average Jitter Last Ten Seconds # - more than 255 ms |
The average jitter in received packets from the last ten one-second intervals. # — maximum (100%) packet loss per second during the one-second interval. |
Packet Loss per Second Last Ten Seconds * - 100% loss |
The 10 recent one-second samples of the lost packet information for the requested endpoint. * — maximum (100%) packet loss per second during the one-second interval. * is displayed when silence suppression is y on the ip-codec-set screen, or when packet loss = 100%. |
Out of Order Count |
A count of the number of out-of-order packets detected during the current connection. |
SSRC Change for Call |
The number of SSRC changes occurring during the current connection. |
Last Rx Sequence No. |
Last received data packet sequence number. |
Last Tx Sequence No. |
Last transmitted data packet sequence number. |
Echo Return Loss |
Loss of the echo signal, relative to the transmitted signal, due to the PSTN network including the hybrid. |
Bulk Delay |
Delay offset of the echo relative to the original signal. |
ERL Enhancement |
Loss introduced by the echo canceller. This is the measure of the effectiveness of the echo canceller. |
Summary |
|
Worst Case this Call |
Jitter: the worst-case, 1-second jitter (ms) experienced during the current connection. Packet Loss: the worst-case, 1-second packet loss experienced during the current connection. |
Average this Call |
Jitter: the average jitter for the current connection (the running average of all the 1-second intervals during the connection. Packet Loss: the average packet loss number for the current connection (running average of all the 1-second intervals experienced during the connection. |
Current Buffer Size |
The current jitter buffer size. |
Connected Ports fields show information for only the IP telephone when an IP telephone and IP Softphone are in service on the same extension simultaneously.
Field |
Description |
|---|---|
src port |
The port address of the statused station. |
MP |
The physical port location of the media processing media module. The audio stream passes through a media processing media module (either a Medpro or a VoIP module on a gateway) when direct IP-IP audio connections are disabled or the connection cannot be direct for other reasons (for example, the call requires a codec conversion). |
HP |
Hairpinning. y is displayed when the audio stream is handled entirely on the media processing media module. The audio connection does not require more extensive processing (for example, some codec conversion), and does not use TDM bus resources. |
ip-start |
The IP address of one end of a direct connection. |
ip-end |
The IP address of the other end of a direct connection. |
audio |
Type of audio codec active for the connection. G.711-MU, G711-A, G.729A, G.723.1-6.3, G.723.1-5.3, PCM, 711-MU, 711-A, 729, 729A, 729B, 729-AB, or 723.1-5.3/6.3. |
encryption |
Type of media encryption as entered on the ip-codec-set screen. aes — Advanced Encryption Standard encryption, standard used by U.S. government to protect sensitive (unclassified) information. Reduces circuit-switched to IP call capacity by 25%. aea — Avaya Encryption Algorithm. Not as secure as AES. none — An unencrypted media stream. |
ss |
on/off Indicates whether silence suppression is active on this audio stream. |
pkt |
Size of the packet carrying the audio. |
dst port |
The port address of one of the other parties to which the statused station has a connection on this call. |
Field |
Description |
|---|---|
src port |
port number |
Field |
Description |
|---|---|
TN Code |
The media processor board code used for the call. |
Board Loc |
The carrier/slot location of the media processor for which data is being reported. |
Codec |
The codec used for the call. |
Encryption |
The type of media encryption used on the call (for example, AES, AEA, SRTP 1, and so on. |
DSP Number |
The number of the DSP on the media processor board used for the call. |
Endpoint ID |
The endpoint ID assigned for the call. |
UDP Port |
The UDP port used by the media gateway or media server for the call. |
Called Number |
The number of the endpoint which received the call (far end). |
Dst Net Reg |
The network region where the destination media gateway or media server is located. |
Far-End IP Addr |
IP address of the device on the far end of the call (called party). |
Echo Canc |
Indicates whether the echo cancellation is on/off for the call. |
Echo Tail (ms) |
The tail length of the echo canceller used for the call. 0 — implies that a different echo canceller was used than the one on the media processor. |
Silence Suppression |
Indicates whether Voice Activation Detection (VAD) is used for the call. |
Comfort Noise Gen |
Indicates whether CNG is being used for the call. |
Data Call/Type |
Indicates whether the call is a data call. If y (yes), include type of data call (for example, FAX, Modem, Clear Channel, TTY, Fax-PT (pass-thru), Mod-PT, TTY-PT). For example, Data Call/Type: y/T.38 FAX |
Threshold Exceptions |
A total number of thresholds that are exceeded for the call when the data was requested. |
Packet Size (ms) |
The size for each steam of data for the associated call, measured in milliseconds. |
% Packet Loss |
The amount of packet loss occurred for the call when the data was requested. |
Peak Packet Loss (%) |
The peak packet loss recorded for the call. |
Jitter Buffer Size (ms) |
The size of the jitter buffer used for the call, measured in milliseconds. |
Jitter Buffer Overruns |
A number of jitter buffer overruns occurred for the call. Overruns occur when too many packets arrive into the jitter buffer very quickly, causing the jitter buffer to fill up. When it happens, the jitter buffer is unable to handle additional traffic/packets. If the number of overruns exceeds 99, the value in this field is 99+. |
Jitter Buffer Underruns |
A number of jitter buffer underruns occurred for the call. When the arrival time of packets goes beyond the size of the jitter butter, the jitter buffer underrun occurs which results in silence until there are additional packets in the jitter buffer to process. If the number of underruns exceeds 99, the value in this field is 99+. |
Average Jitter (ms) |
The average amount of jitter recorded for the call over a 10-second reporting interval, measured in milliseconds. |
Peak Jitter (ms) |
The peak amount of jitter recorded for the call, measured in milliseconds. |
Avg RT Delay (ms) |
The average round trip delay of packets, measured in milliseconds. |
Peak RT Delay (ms) |
The peak round trip delay recorded for the call, measured in milliseconds. |