IP Office

How do i open a config in IP Office?

To do basic administration in IP Office, we would use a GUI application called Manager.  Usually it is installed on the PC running Voicemail Pro.

The default location is “C:All Programs/Avaya/IP Office/Manager/Manager.exe”.

Once you run that program, as it comes up, it will scan the sub-net or look for the specific IP address you entered last time it was used.  If that works, you will see a pop up of your IP Office, or all of your IP Offices.

Select the IP office you desire to make changes to, and click OK in the bottom right corner.  At this point you will be prompted for a username and password.  While the installers might have changed this, the default is: Administrator/Administrator.

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IP Office

How do I save Changes in IP Office Manager

Click on the disk icon (3rd from left on toolbar).

Click OK to upload the new configuration and enter username and password.

*****Note, there will be instances where some of the changes require a reboot for the changes to take effect.  When the reboot window pops up, your choices are:

    • MERGE – Changes saved, no one knows that changes were made (No reboot)
    • IMMEDIATE – Will reboot immediately.  (Will end all calls without warning)
    • WHEN FREE – Will reboot when everyone is off the phone
    • TIMED – You can then set a specific time that the system will reboot.  NOT RECOMMENDED, as you will not know if the reboot happened or if the system comes up again.

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    • IP Office

    How do I change Voicemail Password in IP Office

    To Change Voicemail Passwords: 
    • Open Manager
    • Click on Users
    • Click on the User’s name.
    • Click on the Voicemail tab.
    • Enter temporary password in field.  Or leave blank (this will have no password for the User, so simply press # when prompted for password.)  If you enter a new password, the User will be prompted to change the password when they log in.
    • Click OK when changes are finished.
    • Save the configuration.
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    IP Office

    Backup Voicemail WITH Voicemail to Email

    In a previous post, we chatted about setting a backup Voicemail server.  But, once you do so, because of some requirements, you may break Voicemail to email.  Lets discuss how we can cure this, and do both.

    In setting up the Voicemail Pro Failover in Linux/server Voicemail pro, the syncing SMTP service needs to be the top entry.  The subsequent entries then would be our SMTP to the email server for Voicemail to Email.

    We need to add our entry into the Voicemail Pro client, so open the client and go to: Administration>Preferences>General and go to the EMAIL tab.  Once there, go to the SMTP Sender tab.  To add a new entry, click on the Green Plus sign.

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    IP Office

    Create a moderated conference bridge in IP Office

    One thing that comes up in IP Office is, can the conferencing in Voicemail Pro, make a bridge that users hear hold music until the conference owner arrives?  With conference meet me, when users arrive, they are just brought into the bridge, hearing a beep every time someone new arrives.  But, with release 8, Avaya added the functionality to allow a conference owner to arrive, prior to that arrival, the users in the bridge would hear hold music.  Lets talk about how this can happen.

    Basically, documentation says that if the short code for conference meet me, matches the extension number, this feature is enabled.  So to test that theory, lets look at the differences in short codes.

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    IP Office

    How to set up Windows DHCP Server

    How to configure Windows DHCP for Avaya IP telephones(Compliments of Kyle L Holladay, Sr, R.I.P.)

    In this example we will configure Windows DHCP for Avaya IP telephones. In addition to your standard Option 003 Router you will also need a custom scope option in order for an Avaya IP phone to boot properly using DHCP. While I do reference the IP Office in this document the content is not specific to the IP Office. Options 176 and 242 are common to all Avaya IP telephones and this method would work equally well for phones connected back the an Avaya Aura or Avaya Communication Manager (aka ACM or CM)

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    IP Office

    No User Source Codes

    Standard User Source Numbers

    ValueDescription
    AT<string>Strings beginning with AT can be used with a user called DTEDefault to configure the default settings of the control unit’s DTE port.
    BST_MESSAGE_FOR_YOUIf set, then the BST phone user sees the top line Message for you or Messages for you, indicating that voicemail messages are present. This source number can be used as a NoUser source number to enable the feature for all users.
    BST_NO_MESSAGE_FOR_YOUIf the source number above has been used as a NoUser source number to enable the feature for all BST users, this individual user source number can be used to disable the feature for selected users. If set, the user does not see a message indication when the NoUser setting BST_MESSAGE_FOR_YOU is set. The user’s phone presents the idle date/time in the normal fashion.
    Enable_OTTEnable one touch transfer operation for the user.
    H<Group_Name>Allows the user to receive message waiting indication of new group messages. Enter H followed by the group name, for example HMain. The group is added to the user’s Visual Voice menu.

    On suitable display extensions, the hunt group name and number of new messages is displayed. Refer to the appropriate telephone user guide.

    If the user is not a member of the group, a voicemail code must be set for the group’s mailbox. See the setting Group | Voicemail | Voicemail Code .

    P<Telephone Number>This record sets the destination for callback (outbound alert) calls from voicemail. Enter Pfollowed by the telephone number including any necessary external dialing prefix, for example P917325559876. This facility is only available when using Voicemail Pro through which a default Callback or a user specific Callback start point has been configured. Refer to the Voicemail Pro documentation. This feature is separate from voicemail ringback and Voicemail Pro outcalling.
    R<Caller’s ICLID>To allow Dial In/RAS call access only from a specified number prefix the number with a “R”, for example R7325551234.
    U<User_Name or Extension#>Allows the user to receive message waiting indication of new messages. Enter U followed by the user name or extension number, for example U201. The specified user is added to the user’s Visual Voice menu.

    On suitable display extensions, the user name and number of new messages is displayed. Refer to the appropriate telephone user guide.

    If the user is not a trusted source and a Voicemail Code exists, the user must enter the Voicemail Code corresponding to the monitored mailbox.

    V<Caller’s ICLID>Strings prefixed with a V indicate numbers from which access to the users mailbox is allowed without requiring entry of the mailbox’s voicemail code. This is referred to as “trusted source”.

    For Voicemail Pro running in Intuity mode, trusted source is used for calls from programmable buttons set to Voicemail Collect and Visual Voice. Other controls are prompted for the mailbox number and then password.

    System Wide Source Numbers

    ValueDescription
    ATM4U_PCS7_RINGDETECTFor some cellular or mobile interfaces connected to a IP500 ATM4U
    card, the card may not detect the ring signal. For PCS4 and higher
    card this NoUser source number can be used activate alternate ring
    detection. Refer to IP Office Technical Tip 204.
    ACD_QUEUE_DELAY=nnSoftware level = Up to 3.2.
    Used to change the timeout for still queued messages. The parameter nn can be replace with a time in seconds between 20 and 180. For IP Office 4.0+ this has been replaced by Hunt Group | Announcements.
    ALLOW_5410_UPGRADESPreviously the only control over the upgrading of 5410 phones was
    controlled by the use of the turn_on.bat and turn_off.bat batch files
    installed with the Manager application. Now in addition, this option
    must be present for 5410 phones to update their firmware. Refer to
    the IP Office Installation manual for full details.
    B_DISABLE_SIP_IPADDRDisables the blacklisting of SIP device registration based on the
    device IP address. Refer to the IP Office Security Guidelines document
    for more details.
    BST_MESSAGE_FOR_YOUIf set, all BST phones display the top line Message for you or Messages for
    you    , indicating that voicemail messages are present.
    DECT_REVERSE_RINGBy default, when this parameter is not set, calls on DECT phones
    associated with a CTI application will ring as a Priority call. When
    this parameter is set, DECT phones ring as a normal, external or internal,
    call.
    DISTINCT_HOLD_RINGBACKUsed to display a specific message about the call type for calls
    returning after timing out from being parked or held. If set, such
    calls display Return
    Call – Held     or Return Call – Parked rather than connected party name
    or line name.
     ENABLE_46XXSPECIALS_TXT Loads an additional settings file after the autogenerated 46xxsettings.txt file
    Enable_OTTEnable one touch transfer for all users.
    FORCE_HANDSFREE_TRANSFERIf set, when using the handsfree announced transfer process, both
    the transfer enquiry and transfer completion calls are auto-answered.
    Without this setting only the transfer enquiry call is auto-answered.
    H323SetupTimerNoLCRUsed to set the fallback time from VoIP trunks to non-VoIP trunks within LCR. See IP Trunk Fallback
    HIDE_CALL_STATEUsed to hide the call status information, for example Dial, Conn,
    etc, on DS phones. Used in conjunction with the LONGER_NAMES option.
    Not supported for 1600 and 9600 Series phones.
    IGNORE_DTMF_MISMATCHIPO will ignore DTMF mismatch due to different
    protocols used and allow for direct media call between:
    – H.323 and SIP endpoints
    – DECT and SIP endpoints 
    LONGER_NAMESUsed to increase the length of names sent for display on older
    DS phones, i.e. 2400, 4400 and 5400 Series.
    IPDECT_EXTERNAL_CLI_PRESENTEDAllows incoming CLID to be displayed on the DECT phones
    MEDIA_NAT_DM_INTERNAL=XUsed in conjunction with the setting System
    | VoIP | Allow Direct Media Within NAT Location

    When Allow Direct Media Within NAT Location is set to on, The default behavior is to allow direct media between all types of devices (H323 and SIP remote workers and IP Office Lines behind a NAT). In the case of routers that have H323 or SIP ALG, it can be desirable to allow direct media only between certain categories of devices. In this case, set this NoUser user source number where X is a hex number defined as a  combination of the following flags:

    • 0x01 (includes H323 phones)
    • 0x02 (includes SIP phones)
    • 0x04 (includes IP Office Lines)

    For example, if the router has SIP ALG that can’t be disabled,
    you might want to disable direct media for SIP devices. To configure,
    set MEDIA_NAT_DM_INTERNAL=5 to include only
    H323 phones and IP Office Lines.

    NI2_CALLED_PARTY_PLAN=XX = UNKNOWN or ISDN

    Forces the NI2 Called Party Numbering plan for ETSI PRI trunks.

    NI2_CALLED_PARTY_TYPE=XX = UNKNOWN, INTERNATIONAL, NATIONAL or SUBSCRIBER

    Forces the NI2 Called Party Numbering type for ETSI PRI trunks.

    NI2_CALLING_PARTY_PLAN=XX = UNKNOWN or ISDN

    Forces the NI2 Calling Party Numbering plan for ETSI PRI trunks.

    NI2_CALLING_PARTY_PLAN=XX = UNKNOWN, INTERNATIONAL, NATIONAL or SUBSCRIBER

    Forces the NI2 Calling Party Numbering type for ETSI PRI trunks.

    NO_DIALLED_REF_EXTERNALOn outgoing external calls made using short codes to dial the full
    number, only the short code dialed is displayed on the dialing user’s
    phone and any directory matching is based on that number dialled.
    On systems with this source number added to the configuration, after
    dialing a short code the full number dialled by that short code is
    shown and directory matching is based on that full number.
    onex_l1=XSets the IP address of the one-X server that can be accessed by
    clients registered on the LAN1 interface.
    onex_l2=XSets the IP address of the one-X server that can be accessed by
    clients registered on the LAN2 interface.
    onex_port_l1=XSets the port of the one-X server that can be accessed by clients
    registered on the LAN1 interface.
    onex_port_l2=XSets the port of the one-X server that can be accessed by clients
    registered on the LAN2 interface.
    onex_port_r1=XSets the port of the one-X server that can be accessed by remote
    clients registered on the LAN1 interface.
    onex_port_r2=XSets the port of the one-X server that can be accessed by remote
    clients registered on the LAN2 interface.
    onex_r1=XSets the IP address of the one-X server that can be accessed by
    remote clients registered on the LAN1 interface.
    onex_r2=XSets the IP address of the one-X server that can be accessed by
    remote clients registered on the LAN2 interface.
    PATCHES 0x0020 allows telnet access
    PRESERVED_CONN_DURATION=XX = time in minutes. Range = 1 to 120.

    When the setting System
    | Telephony | Telephony | Media Connection Preservation     is enabled, preserved calls have a maximum
    duration of 120 minutes. After that time, they are hung up. Use this
    setting to change the maximum duration value.

    PRESERVED_NO_MEDIA_DURATION=XX = time in minutes. Range = 1 to 120.

    When the setting System
    | Telephony | Telephony | Media Connection Preservation     is enabled, preserved calls have a maximum
    duration of 120 minutes. If monitoring RTP or RTCP and no speech is
    detected, calls are hung up after 10 minutes. Use this setting to
    change the default value of 10 minutes.

    ProgressEndsOverlapSendSee Line | VoIP.
    REPEATING_BEEP_ON_LISTENBy default, if you set Beep on Listen and invoke Call Listen you’ll
    hear an entry tone (3 beeps). When this parameter is set, you hear
    a beep every 10 seconds when you invoke Call Listen.
    REMOTE_H323=1800Register H.323 phones via port 1800
    RTCP_COLLECTOR_IP=XX = IP address of the IP Office system as configured in the Prognosis
    server.
    RW_SBC_REG=<SBC-B1-public-SIP-IPaddr>Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
    on 11xx, 12xx, and E129 phones. The IP address is used as a S1/S2
    for 11xx and 12xx and for outbound-proxy-server for E129 sets.
    RW_SBC_PROV=<SBC-B1-private-HTTP/S-IPaddr>Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
    on 11xx, 12xx, and E129 phones. The IP address is used to determine
    whether a 11xx, 12xx, or E129 set is registered as an IP
    Office SBCE Remote Worker. 
    RW_SBC_TLS=<SBC-public-TLS-port>Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
    on 11xx, 12xx, and E129 phones. The  port is used as a S1/S2 TLS
    port for 11xx and 12xx phones and as outbound-proxy-server TLS port
    for E129 phones.
    RW_SBC_TCP=<SBC-public-TCP-port>Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
    on 11xx, 12xx, and E129 phones. The port is used as a S1/S2 TCP
    port for 11xx and 12xx phones and as outbound-proxy-server TCP port
    for E129 phones.
    RW_SBC_UDP=<SBC-public-UDP-port>Used for Remote Worker Session Boarder Controller Enterprise (SBCE) configuration
    on 11xx, 12xx, and E129 phones. The port is used as a S1/S2 UDP
    port for 11xx and 12xx phones and as outbound-proxy-server UDP port
    for E129 phones. 
    SET_46xx_PROCPSWD=XX= New password

    When set, the new password is indicated to phones through the auto-generated
    settings file.

    SET_96xx_SIG=XWhen set, inserts the line “SET SIG X into the auto-generated
    settings files.
    SET_HEADSYS_1If set, alters the operation of the headset button on 96×1 phones
    via the auto-generated settings file. Normally the headset goes off-hook
    when the far end disconnects. When this option is set, the headset
    remains on-hook when the far end disconnects.
    SHOW_LINEID_NOT_OUTSIDEBy default, for calls where no incoming caller ID (ICLID) information is available, the IP Office inserts the word “External” wherever 
    ICLID information is normally displayed. The NoUser source number value SHOW_LINEID_NOT_OUTSIDE can be used to make available within 
    the configuration for each trunk and channel Line Name and Channel Name fields. The text entered into those fields is then used 
    with external calls without ICLID information.o This feature is not used with SIP, IP DECT, E1R2 and S0 lines.
    o On T1 lines, a Name field is also made available for individual channels and if set overrides the line name field.
    o This feature does not override the display of Withheld if the caller has withheld their ICLID information.
    o Where a Name is entered, that value is used to identify calls with no ICLID information.
    o For line appearance buttons, if set the Name replaces the Line Appearance ID as the default button label.
    SIP_E129_PREFER_UDPWhen set, the auto-generated E129 configuration file is altered
    to set the transport method as UDP regardless of whether TCP or TLS
    are selected on the LAN1/LAN2 VoIP configuration settings.
    SIP_ENABLE_HOT_DESKFor IP Office Release 10.1, by default the use of hot-desking on
    J129 and H175 phones is blocked. This source numbers overrides that
    behavior.
    SIP_LINE_NEAR_HOLD=N Where N is the active SIP Line number.  With this NUSN string configured in IP Office, both RTP and RTCP packets are sent from IP Office to SIP trunk when on-hold resolving the call-hold issue.
    SIP_EXTN_CALL_Q_TIMEOUT=XX = Number of minutes (0 (no limit) to 255).

    Sets the unanswered call duration after which unanswered SIP calls
    are automatically disconnected. If not set, the normal default is
    5 minutes.

    SIP_OPTIONS_PERIOD=XX = time in minutes. The system sends SIP options messages periodically
    to determine if the SIP connection is active. The rate at which the
    messages are sent is determined by the combination of the Binding Refresh Time (in seconds) set on the Network Topology tab and the SIP_OPTIONS_PERIOD parameter (in minutes). The frequency of sent messages is determined
    as follows:

    If no SIP_OPTIONS_PERIOD parameter is defined and the Binding Refresh Time is 0, then
    the default value of 300 seconds is used.

    To establish a period less than 300 seconds, do not define a SIP_OPTIONS_PERIOD parameter and set the Binding Refresh Time to a value less than 300 seconds. The OPTIONS message period will
    be equal to the Binding Refresh Time.

    To establish a period greater than 300 seconds, a SIP_OPTIONS_PERIOD parameter must be defined. The Binding Refresh Time must be set to a value greater than 300 seconds. The OPTIONS message
    period will be the smaller of the Binding Refresh Time and the SIP_OPTIONS_PERIOD.

    SOFTPHONE_RTP_MAX=XX = Maximum port in the range 1024 to 65534.

    The maximum usable port indicated to the IP Office Video Softphone
    when SOFTPHONE_RTP_RANGE_ENABLE and SOFTPHONE_RTP_MIN are set.

    SOFTPHONE_RTP_MIN=XX = Minimum port in the range 1024 to 65534.

    The minimum usable port indicated to the IP Office Video Softphone
    when SOFTPHONE_RTP_RANGE_ENABLE and SOFTPHONE_RTP_MAX are set.

    SOFTPHONE_RTP_RANGE_ENABLEWhen set, the usable ports indicated to the IP Office Video Softphone
    are set via the SOFTPHONE_RTP_MIN and SOFTPHONE_RTP_MAX values.
    SUPPRESS_ALARM=1When set, suppresses the NoCallerID alarm otherwise shown in SysMonitor,
    SNMP traps, email notifications, SysLog or System Status.
    TUI:NAME_SEARCH_MODE=1The default directory search matching behavior is to simultaneously
    match against first and last name characters. This source number sets
    the system to match from the start of the name only.
    TAPI_SPECIAL_LLC IPOCC Specific
    TAPI_REPORTS_TWIN_CALLSMake sure users with twinning show up as busy in tapi when the have a twinned call
    VM_TRUNCATE_TIME=XX= time in seconds. Range = 0 to 7.

    On analog trunks, call disconnection can occur though busy tone
    detection. When such calls go to voicemail to be recorded or leave
    a message, when the call ends the system indicates to the voicemail
    server how much to remove from the end of the recording in order to
    remove the busy tone segment. This amount varies by system locale,
    the defaults being listed below. For some systems it may be necessary
    to override the default if analog call recordings are being clipped
    or include busy tone. That can be done by adding a VM_TRUNCATE_TIME= setting with the required value in the range 0 to 7 seconds.

    • New Zealand, Australia,
      China, Saudi Arabia and Custom      : 5 seconds.

    • Korea: 3 seconds.

    • Italy, Mexico, Chile,
      Colombia and Brazil      : 2 seconds.

    • Argentina, United States,
      Canada and Turkey      : 0 seconds.

    • All other locales: 7 seconds.

    VMAIL_WAIT_DURATION=XThe number of milliseconds to wait before cutting through the audio
    to Voicemail. Some delay is required to allow for codec negotiation.
    VMPRO_OOB_DTMF_OFFWhen set, disabled the sending of out-of-band digits to the Voicemail
    Pro voicemail server.
    xmpp_port_l1=XX = The port of the XMPP server that can be accessed by clients
    registered on the LAN1 interface.
    xmpp_port_l2=XX = The port of the XMPP server that can be accessed by clients
    registered on the LAN2 interface.
    xmpp_port_r1=XX = The port of the XMPP server that can be accessed by remote
    clients registered on the LAN1 interface.
    xmpp_port_r2=XX = The port of the XMPP server that can be accessed by remote
    clients registered on the LAN2 interface.

    Also another tip, from @Intrigant…

    SCN Calls forced over PSTN
    Create an ARS naming **NET**
    Put in the shortcodes you need.
    Set the scn channel to 0

    Now it worked like pstn overflow. 
    Only signaling is used over scn.

    Update from Sizbut:

    By default, third-party SIP phones support up to 6 simultaneous calls. However, using a user source number, you can increase this to 30 calls.

    The user Source Number of ULI=N allows a third-party SIP extension to consume multiple third-party endpoint licenses, where N is the number of additional licenses from 1 to 4.

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    IP Office

    Setting up One-X Portal for the Preferred Mobility Application

    A common request is “How do I set up the Preferred Mobility client”? The intent of this guide is to show the necessary steps to configure. Please refer to the Avaya One-X Mobile Preferred for IP Office guide for complete installation information.

    Setting up the new One-X Mobile Preferred app is easy and straight-forward. If the One-X Portal server will be behind a firewall, TCP port forwarding will need to be set up.

    Setting Up Mobile Preferred Client

    1.) The first step is to create an FQDN (definition here). The app needs to connect to the FQDN via 3G or wifi service.

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    Merlin Legend/Magix

    Labeling Extensions, Trunks, or Speed dials on Merlin Magix

    Through the use of the Labeling feature, the system manager can program the system to provide identification information (labels and posted messages) on display telephones.

    Alphanumeric labels can be assigned to the following:

    * System Speed Dial Directory – Use this procedure to establish System Speed Dial numbers for all system users. You can also use this procedure to enter the alphanumeric labels shown on display telephones (for the System Directory feature of the MLX telephone).

    * Extension Directory Listings – Use this procedure to establish alphanumeric system labels for display telephone users to identify the person calling or leaving a message. This procedure is also used to program the Extension Directory feature for MLX telephones.

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