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Avaya Aura® Communication Manager Screen Reference
IP options system parameters
Last Updated : Oct 29, 2012 |
Example command: change system-parameters ip-options
AUTOMATIC TRACE ROUTE ON
Link Failure
Enables or disables the automatic trace route command. If enabled, to diagnose network problems, especially to determine where a network outage exists, Communication Manager initiates an automatic trace-route command when the connectivity between a server and its gateways, or IP trunks is lost.
Note:
If disabled, any automatic trace-route currently in progress finishes, and no subsequent trace-route commands are started or logged. In other words, the link failure buffer is cleared.
H.248 MEDIA GATEWAY
Link Loss Delay Timeout (minutes)
Valid Entry |
Usage |
|---|---|
1 to 30 |
The number of minutes to delay the reaction of the call controller to a link bounce. Assists with the H.248 link bounce recovery mechanism of the Avaya G700 branch gateway. Specifically, prevents the call controller from removing all boards and ports prematurely in response to a link bounce. Default is 5. |
H.323 IP ENDPOINT
Link Loss Delay Timer (minutes)
Valid Entry |
Usage |
|---|---|
1 to 60 |
The number of minutes to delay the reaction of the call controller to a link bounce. Specifies how long the Communication Manager server preserves registration and any stable calls that might exist on the endpoint after it has lost the call signaling channel to the endpoint. If the endpoint does not re-establish connection within this period, the system cancels the registration and any calls of the endpoint. This timer does not apply to soft IP endpoints operating in telecommuter mode. Default is 5. |
Periodic Registration Timer (min)
This timer is started when an IP telephone registration is taken over by another IP endpoint. When the timer expires, the telephone tries to reregister with the server. Default timer value is dependent on the number of unsuccessful periodic registration attempts. Sample field values apply unless the endpoint is interrupted, such as by power loss, or the user takes manual action to override this automatic process:
20 means once every 20 minutes for two hours, then once an hour for 24 hours, then once every 24 hours continually.
60 means once an hour for two hours, then once an hour for 24 hours, then once every 24 hours continually.
Valid Entry |
Usage |
|---|---|
1 to 60 |
The number of minutes before an IP telephone registration is taken over by another IP endpoint attempts to re-register with the server. Default is 60. |
Primary Search Time (seconds)
Valid Entry |
Usage |
|---|---|
15 to 3600 |
The maximum number of seconds the IP endpoint attempts to register with its current Communication Manager server while the telephone is disconnected before going to a Survivable Remote Server. With this timer, the customer can specify the maximum time that an IP endpoint spends on trying to connect to the PROCR before going to a Survivable Remote Server. When the IP telephone’s receiver is lifted, the endpoint continues trying to re-establish connection with the current server until the call ends. Default is 75. |
Short/Prefixed Registration Allowed
The system displays Short/Prefixed Registration Allowed if the IP Stations field on the System Parameters Customer Options screen is set to y.
Valid Entry |
Usage |
|---|---|
y |
Call Processing allows an IP endpoint to register using a short extension, for the extensions that have Short/Prefixed Registration Allowed field set to default on the Station screen. The default value is y. |
n |
Call Processing does not allow an IP endpoint to register using a short extension, for the extensions that have Short/Prefixed Registration Allowed field set to default on the Station screen. |
IP MEDIA PACKET PERFORMANCE THRESHOLDS
Enable Voice/Network Stats
Enables or disables the recording of voice and network statistics at a system level for all TN2302/TN2602 media processor boards in the network. The default value is disabled.
Number of Pings Per Measurement Interval
Valid Entry |
Usage |
|---|---|
10 to 100 |
the number of test pings that comprise a measurement from which the performance values (delay and loss) are calculated. Default is 10. |
Packet Loss (%)
Specifies thresholds to be applied to packet loss rates (as measured by ping) for determining activation or deactivation of signaling group bypass.
High
Valid Entry |
Usage |
|---|---|
0 to 100 |
The high value cannot be less than the minimum value. Default is 40. |
Low
Valid Entry |
Usage |
|---|---|
0 to 100 |
The low value cannot be more than the maximum value. Default is 15. |
Ping Test Interval (sec)
Valid Entry |
Usage |
|---|---|
10 to 999 |
The time between performance test pings for each testable signaling group. Default is 20. |
Roundtrip Propagation Delay (ms)
Specifies thresholds to be applied to roundtrip packet propagation delays as measured by ping, for use in activating or clearing signaling group bypass.
High
Valid Entry |
Usage |
|---|---|
10 to 9999 |
The high value cannot be less than the minimum value. Default is 800. |
Low
Valid Entry |
Usage |
|---|---|
10 to 9999 |
The low value cannot be more than the maximum value. Default is 400. |
MEDIA GATEWAY ANNOUNCEMENT SERVER PARAMETERS
Announcement Server IP Address
The IP address of the Announcement Server that is a unique IP address assigned to each port on any IP device that is used for a connection.
Announcement Storage Path Name
The directory path name on the Announcement Server where the announcements are stored. Accepts up to 40 characters.
Login
The login used by the gateway to access the announcement server. Accepts up to 10 characters.
Password
The password used by the gateway to access the announcement server. Accepts up to 10 characters.
RTCP MONITOR SERVER
The RTCP monitor is a separate computer that receives RTCP packets from many devices. Communication Manager pushes these values to IP telephones, IP softphones and VoIP media modules, such that they know where to send the data.
Default RTCP Report Period (secs)
Valid Entry |
Usage |
|---|---|
5 to 99 |
The number of seconds IP telephones, IP softphones, and VoIP media modules send RTCP packets to the RTCP server. |
Default Server IP Address
The default IP address of the RTCP server used for each administered region. A unique IP address is assigned to each port on any IP device that is used for a connection.
Default Server Port
Valid Entry |
Usage |
|---|---|
1 to 65535 |
The default TCP/IP port of the RTCP server. Default is 5005. |
Link Failure
Enables or disables the automatic trace route command. If enabled, to diagnose network problems, especially to determine where a network outage exists, Communication Manager initiates an automatic trace-route command when the connectivity between a server and its gateways, or IP trunks is lost.
Note:
If disabled, any automatic trace-route currently in progress finishes, and no subsequent trace-route commands are started or logged. In other words, the link failure buffer is cleared.
Prefer use of G.711 by Music Sources
Enables or disables using G.711 for intra-switch Music-On-Hold. The default value is n.
Force Phones and Gateways to Active LSPs
Enables or disables forcing telephones and gateways to active LSPs. The default is disabled.
HYPERACTIVE MEDIA GATEWAY REGISTRATIONS
Enable Detection and Alarms
Enables or disables the hyperactive gateway registration feature. Default is disabled.
Number of Registrations within the Window
Valid Entry |
Usage |
|---|---|
1 to 19 |
The number of registrations that occur within the hyperactivity window for generating a Gateway alarm. Default is 3. Available only if detection and alarming is enabled. |
Parameters for Media Gateway Alarms: Hyperactive Registration Window (minutes)
Available only if detection and alarming is enabled.
Valid Entry |
Usage |
|---|---|
1 to 15 |
The time in minutes for checking hyperactive gateway registrations. Default is 4 minutes. |
Parameters for Network Region Registration (NR-REG) Alarms: % of Gateways in Network Region with Hyperactive Registration Alarms
Available only if detection and alarming is enabled.
Valid Entry |
Usage |
|---|---|
1 to 99 |
The percent of Gateways within an ip-network region that should be alarmed before an IP-Registration alarm is generated. Default is 80%. |
IP DTMF TRANSMISSION MODE
Inter-System IP DTMF Transmission Mode
Specifies the touchtone signals that are used for dual-tone multifrequency (DTMF) telephone signaling.
Valid Entry |
Usage |
|---|---|
in-band |
All G711 and G729 calls pass DTMF in-band. DTMF digits encoded within existing RTP media stream for G.711/G.729 calls. G.723 is sent out-of-band. It also supports the SIP trunks. |
in-band-g711 |
Only G711 calls pass DTMF in-band. The system displays the in-band-g711 option if the Group Type field is set to h.323. |
out-of-band |
All IP calls pass DTMF out-of-band. For IP trunks, the digits are done with either Keypad IEs or H245 indications. This is the default for newly added H.323 signaling groups. The out-of-band option enables out of band signaling for the SIP Signaling group. When you select the out-of-band option, Communication Manager sends all outgoing DTMF messages as SIP INFO messages over a SIP signaling group. This option is interoperable with H.323 networks. You can connect an Avaya non-SIP endpoint or trunk to a voice mail system on the H.323 network that is integrated with Session Manager and Communication Manager through SIP. Examples of Avaya non-SIP endpoints or trunks are H.323, Analog, DCP, and Digital trunks. |
rtp-payload |
This is the method specified by RFC 2833. This is the default value for newly added SIP signaling groups. Support for SIP trunks requires the default entry of rtp-payload. |
Intra-System IP DTMF Transmission Mode
The IP transmission mode.
Valid Entry |
Usage |
|---|---|
in-band |
DTMF digits encoded within existing RTP media stream for G.711/G.729 calls. G.723 is sent out-of-band. |
rtp-payload |
Support for SIP trunks requires the entry of rtp-payload. |
ALTERNATIVE NETWORK ADDRESS TYPES
To control the call processing behavior to send Alternative Network Address Types (ANAT) offer system wide.
ANAT Enabled
Use this field to control the call processing behavior to send Alternative Network Address Types (ANAT) offer system wide.
Valid entry |
Usage |
|---|---|
y |
Communication Manager sends ANAT offer to the SIP elements system wide. |
n |
Communication Manager does not send ANAT offer to the SIP elements system wide. |
SERVICES DIALPAD PARAMETERS
To disable the services dialpad parameters on the telephones, set the Download Flag field to y and leave the Password field blank.
Download Flag
If you set this field to y, Communication Manager downloads the services dialpad parameters and the associated IP addresses to the telephones. The default value is n.
Password
Use this field to enter the password that a technician uses to administer craft procedures. During the installation or after the successful installation of an IP telephone, the technician performs craft procedures to customize the telephone installation for specific operating environment. You can provide a password of length up to seven digits or leave the field blank. The default value is 27238.
SNMP STATION PARAMETERS
To disable the SNMP parameters on the telephones, set the Download Flag field to y and leave the Community String field blank.
Community String
Use this field to enter a string that an IP telephone uses to determine whether the telephone must receive SNMP messages. You can enter up to 32 characters.
The telephone does not respond to the incoming SNMP message if you:
leave the Community String field blank and the source address of the SNMP message does not match with an address in the SNMP Source Address list
leave the Community String field blank and the source address of the SNMP message matches with an address in the SNMP Source Address list
set the Community String field to a value and the source address of the SNMP message does not match with an address in the SNMP Source Address list
The telephone responds to the incoming SNMP message only if you set the Community String field to a value and the source address of the SNMP message matches with an address in the SNMP Source Address list.
Download Flag
If you set this field to y, Communication Manager downloads the SNMP parameters and the associated IP addresses to the telephones. The default value is n.
SOURCE ADDRESSES
Use this field to enter node names, mapping to proper IP address, that an IP telephone uses to validate the source address of an SNMP message. The telephone does not respond to the incoming SNMP message if you:
leave the Community String field blank and the source address of the SNMP message does not match with an address in the SNMP Source Address list
leave the Community String field blank and the source address of the SNMP message matches with an address in the SNMP Source Address list
set the Community String field to a value and the source address of the SNMP message does not match with an address in the SNMP Source Address list
The telephone responds to the incoming SNMP message only if you set the Community String field to a value and the source address of the SNMP message matches with an address in the SNMP Source Address list.
You can enter up to six node names. Use the IP Node Names screen to administer node names.
Dest # 1, 2, or 3 IP address
The valid destination IPv4 address. The default destination address is 0.0.0.0.
Local Facility
Valid Entry |
Usage |
|---|---|
local0 to local7 |
Displays the help message upon acceptable values for local use. The default value is local4. |
Port #
Valid Entry |
Usage |
|---|---|
1 to 65535 |
The valid port number associated with the destination IPv4 address. The default port number is 514. |
MUSIC/ANNOUNCEMENTS IP-CODEC PREFERENCES
In general, when operating across the WAN with limited bandwidth facilities, the ip-codec-set is configured only with the compressed voice codecs. Sometimes it is necessary to carry the voice calls with the normal configured compressed codecs, and the music or announcement sources are received with non-compressed G.711 codecs.
If you administer the following fields, when possible, the system overrides the ip-codec-set preference which can be configured to prefer a compressed codec, with non-compressed G.711. If the device receives the music or announcement source which has signaled support for G.711, the system attempts to use G.711.
Prefer use of G.711 by Announcement Sources
Overrides the ip-codec-set preference and establishes inter-PN or inter-gateway connections transmitting announcements with G.711. The default value is n.
Prefer use of G.711 by IP Endpoints Listening to Announcements
Overrides the ip-codec-set preference and reconfigures the IP endpoints listening to an announcement source with G.711. The default value is n.
Prefer use of G.711 by IP Endpoints Listening to Music
Overrides the ip-codec-set preference and reconfigures the IP endpoints listening to a music source with G.711. The default value is n.
Prefer use of G.711 by Music Sources
Overrides the ip-codec-set preference and establishes inter-PN or inter-gateway connections transmitting music with G.711. The default value is n.
BW Management Option
Use this field to select the bandwidth management mode.
Valid entries |
Usage |
|---|---|
local-CM |
With this mode, Communication Manager acts as stand-alone bandwidth management entity. Bandwidth limits for the bandwidth used by Communication Manager are set on Communication Manager. The default value is local-CM. |
shared-SM |
With this mode, Session Manager acts as the central authority for bandwidth management and Communication Manager obtains bandwidth for voice and multimedia IP connections from Session Manager. Bandwidth limits for the bandwidth collectively used by Communication Manager and other users are set using System Manager. For more information, see Administering Avaya Aura® Session Manager. |
Signaling-group of Primary SM BW Mgr
Use this field to specify the assigned Session Manager SIP signaling group number. You can enter a value from 1 to 999. Do not leave the field blank.
The system displays the field only when you set the BW Management Option field to shared-SM.
Signaling-group of Secondary SM BW Mgr
Use this field to specify the assigned Session Manager SIP signaling group number. You can enter a value from 1 to 999. You can leave the field blank. This optional secondary Session Manager SIP signaling group is only used when the primary Session Manager SIP signaling group is unavailable. The secondary Session Manager SIP signaling group cannot be the same number as the primary Session Manager SIP signaling group.
The system displays the field only when you set the BW Management Option field to shared-SM.
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