Standard User Source Numbers
| Value | Description |
|---|---|
| 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_YOU | If 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_YOU | If 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_OTT | Enable 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
| Value | Description |
|---|---|
| ATM4U_PCS7_RINGDETECT | For 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=nn | Software 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_UPGRADES | Previously 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_IPADDR | Disables 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_YOU | If set, all BST phones display the top line Message for you or Messages for you, indicating that voicemail messages are present. |
| DECT_REVERSE_RING | By 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_RINGBACK | Used 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_OTT | Enable one touch transfer for all users. |
| FORCE_HANDSFREE_TRANSFER | If 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. |
| H323SetupTimerNoLCR | Used to set the fallback time from VoIP trunks to non-VoIP trunks within LCR. See IP Trunk Fallback |
| HIDE_CALL_STATE | Used 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_MISMATCH | IPO 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_NAMES | Used to increase the length of names sent for display on older DS phones, i.e. 2400, 4400 and 5400 Series. |
| IPDECT_EXTERNAL_CLI_PRESENTED | Allows incoming CLID to be displayed on the DECT phones |
| MEDIA_NAT_DM_INTERNAL=X | Used 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:
For example, if the router has SIP ALG that can’t be disabled, |
| NI2_CALLED_PARTY_PLAN=X | X = UNKNOWN or ISDN Forces the NI2 Called Party Numbering plan for ETSI PRI trunks. |
| NI2_CALLED_PARTY_TYPE=X | X = UNKNOWN, INTERNATIONAL, NATIONAL or SUBSCRIBER Forces the NI2 Called Party Numbering type for ETSI PRI trunks. |
| NI2_CALLING_PARTY_PLAN=X | X = UNKNOWN or ISDN Forces the NI2 Calling Party Numbering plan for ETSI PRI trunks. |
| NI2_CALLING_PARTY_PLAN=X | X = UNKNOWN, INTERNATIONAL, NATIONAL or SUBSCRIBER Forces the NI2 Calling Party Numbering type for ETSI PRI trunks. |
| NO_DIALLED_REF_EXTERNAL | On 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=X | Sets the IP address of the one-X server that can be accessed by clients registered on the LAN1 interface. |
| onex_l2=X | Sets the IP address of the one-X server that can be accessed by clients registered on the LAN2 interface. |
| onex_port_l1=X | Sets the port of the one-X server that can be accessed by clients registered on the LAN1 interface. |
| onex_port_l2=X | Sets the port of the one-X server that can be accessed by clients registered on the LAN2 interface. |
| onex_port_r1=X | Sets the port of the one-X server that can be accessed by remote clients registered on the LAN1 interface. |
| onex_port_r2=X | Sets the port of the one-X server that can be accessed by remote clients registered on the LAN2 interface. |
| onex_r1=X | Sets the IP address of the one-X server that can be accessed by remote clients registered on the LAN1 interface. |
| onex_r2=X | Sets 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=X | X = time in minutes. Range = 1 to 120. When the setting System |
| PRESERVED_NO_MEDIA_DURATION=X | X = time in minutes. Range = 1 to 120. When the setting System |
| ProgressEndsOverlapSend | See Line | VoIP. |
| REPEATING_BEEP_ON_LISTEN | By 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=1800 | Register H.323 phones via port 1800 |
| RTCP_COLLECTOR_IP=X | X = 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=X | X= New password When set, the new password is indicated to phones through the auto-generated |
| SET_96xx_SIG=X | When set, inserts the line “SET SIG X into the auto-generated settings files. |
| SET_HEADSYS_1 | If 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_OUTSIDE | By 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_UDP | When 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_DESK | For 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=X | X = Number of minutes (0 (no limit) to 255). Sets the unanswered call duration after which unanswered SIP calls |
| SIP_OPTIONS_PERIOD=X | X = 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 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 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 |
| SOFTPHONE_RTP_MAX=X | X = Maximum port in the range 1024 to 65534. The maximum usable port indicated to the IP Office Video Softphone |
| SOFTPHONE_RTP_MIN=X | X = Minimum port in the range 1024 to 65534. The minimum usable port indicated to the IP Office Video Softphone |
| SOFTPHONE_RTP_RANGE_ENABLE | When 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=1 | When set, suppresses the NoCallerID alarm otherwise shown in SysMonitor, SNMP traps, email notifications, SysLog or System Status. |
| TUI:NAME_SEARCH_MODE=1 | The 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_CALLS | Make sure users with twinning show up as busy in tapi when the have a twinned call |
| VM_TRUNCATE_TIME=X | X= time in seconds. Range = 0 to 7. On analog trunks, call disconnection can occur though busy tone
|
| VMAIL_WAIT_DURATION=X | The number of milliseconds to wait before cutting through the audio to Voicemail. Some delay is required to allow for codec negotiation. |
| VMPRO_OOB_DTMF_OFF | When set, disabled the sending of out-of-band digits to the Voicemail Pro voicemail server. |
| xmpp_port_l1=X | X = The port of the XMPP server that can be accessed by clients registered on the LAN1 interface. |
| xmpp_port_l2=X | X = The port of the XMPP server that can be accessed by clients registered on the LAN2 interface. |
| xmpp_port_r1=X | X = The port of the XMPP server that can be accessed by remote clients registered on the LAN1 interface. |
| xmpp_port_r2=X | X = 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.
Each additional license enables another 6 calls, up to a maximum of 30 calls total (4 additional licenses).
For subscription systems, the source number consumes additional user subscriptions.
Changes to the user Source Number require a system restart.
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