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.

June 10, 2013July 16, 2014IP 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)

March 3, 2013September 1, 2023IP Office

No User Source Codes

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 `P`followed 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: 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=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 \| 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=X	X = 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.
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 settings file.
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 are automatically disconnected. If not set, the normal default is 5 minutes.
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 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=X	X = 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=X	X = 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_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 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=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.

February 28, 2013July 18, 2014IP 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.

February 28, 2013May 18, 2014IP Office

How do i route a call in IP Office?

If you need to set up a number to ring in to a specific destination, you set the number in the “Incoming call route”

Open Manager.
Click on the Incoming Call Route tab.
Select an Open DID number. (Or add a new number)
Select the Destination tab.
Click OK when changes are finished.

February 28, 2013May 18, 2014IP Office

Manually set time for Avaya IP Office

Manually Setting the System Time in IP Office

For IP Office systems without access to a time server, a number of methods to manually set the system time exist. These are available in the IP Office 4.1+. The following method is only supported on these phones: 2410, 2420, 4412, 4424, 4612, 4624, 4610, 4620, 4621, 5410, 5420, 5610, 5620, 5621, 6412, 6424. It is supported on any analog, T3, T3 IP or wireless (IP DECT, 3600, 3700 Series) phones.

Note: If the IP Office system that is still configured for automatic time updates (on by default), any manually set time and date will be overridden on the next automatic update. The use of automatic updates can be disabled by setting the System | System | Time Server IP Address to 0.0.0.1.

February 28, 2013May 18, 2014IP Office

8.1 Server Edition

Looking at the new Server Edition, most installers are a bit “nervous”. Relax, there is nothing to be nervous about. In its most simplistic form, there are only 3 (new) parts to 8.1 Server Edition.

First, you need to have a server. The 2 options are:

HP DL360G7 – 12 Gig RAM, 300 Gig Raid drive(dual), dual hot swappable power supplies, and dual quad core processors (2.4Ghz).

HP DL120G7 – 12 Gig RAM, 250 Gig hard drive (single), single power supply, single quad core processor (3.1 Ghz)

February 28, 2013July 9, 2014IP Office

Setting up SalesForce.com plugin for Avaya One-X Portal

Some quick requirements for the Salesforce.com plugin

Operating System: Windows XP or Windows 7 (32-bit and 64-bit).
Salesforce.com CTI toolkit 3.0.
Microsoft .NET 4.0 framework.

Supported Web Browsers:
a. Windows XP: Internet Explorer 7 or Internet Explorer 8 (32-bit).
b. Windows 7 (32-bit): Internet Explorer 8 (32-bit).
c. Windows 7 (64-bit): Internet Explorer 8 (64-bit).

Note: The only supported web browser for using Avaya IP Office Plug-in for Salesforce.com is Internet Explorer 7 or Internet Explorer 8 (32-bit and 64-bit).

Avaya IP Office Plug-in for Salesforce.com works with the following Salesforce Editions:
a. Professional Edition (with API access enabled).
b. Enterprise Edition.
c. Unlimited Edition.
d. Developer Edition.

February 28, 2013May 18, 2014IP Office

How to VPN enable 96XX Phones

This is an FAQ how to enable the vpn software and disable the vpn software on a 96xx phone. This was plagiarized from Peter Ista’s post on Tek-Tips.

The vpn software is different from the 56xx series which needs special vpn firmware.
The 96xx phones do not need any other firmware but you just need to enable it.
Disabling the vpn option is also possible but removing is not a default option.

Here’s how to enable,disable and remove it.

ENABLE THE VPN OPTION:

Open the 46xxsettings.txt and add the next lines:

IF $GROUP SEQ 0 GOTO NO96XXVPN 
IF $GROUP SEQ 876 GOTO 96XXVPN 
# 96XXVPN 
GET 96xxvpn.txt 
# NO96XXVPN 
SET NVVPNMODE 0 
SET VPNPROC 0

Then make a new file called: 96xxvpn.txt

Put in the next lines and change as needed:

################################################## # ## VPN Mode ## 0: Disabled, 1: Enabled. ################################################## #SET NVVPNMODE 1################################################## # ## Vendor. ## 1: Juniper/Netscreen, 2. Cisco ## 3: CheckPoint/ Nokia 4: Other ## 5: Nortel. ################################################## #SET NVVPNSVENDOR 1################################################## # ## Encapsulation Type. ## 0: 4500-4500, 1: Disabled ## 2: 2070-500, 3: ? ## 4: RFC (500-500) ################################################## #SET NVVPNENCAPS 0################################################## # ## Copy TOS. ## 1: Yes, 2: No ################################################## #SET NVVPNCOPYTOS 2 ################################################## # ## Authentication Type. ## ## [For Cisco/Juniper/Checkpoint/Other] ## 3: PSK, 4: PSK with Xauth ## 5: RSA signatures with Xauth, 6: Hybrid Xauth ## 7: RSA signatures. ## ## [Nortel Authentication Type] ## 1: Local credentials, 2: Radius Credentials. ## 3: Radius SecureID, 4: Radius Axent. ################################################## #SET NVVPNAUTHTYPE 5 ################################################## # ## VPN User Type. ## 1: Any, 2: User ################################################## #SET NVVPNUSERTYPE 1 ################################################## # ## VPN User name. ################################################## #SET NVVPNUSER mscep1 ################################################## # ## Password Type. ## 1: Save in Flash, 2: Erase on reset ## 3: Numeric OTP, 4: Alpha-Numeric OTP ## 5: Erase on VPN termination. ################################################## #SET NVVPNPSWDTYPE 1 ################################################## # ## User Password. ################################################## #SET NVVPNPSWD mscep1 ################################################## # ## IKE ID (Group Name). ################################################## #SET NVIKEID mscep ################################################## # ## IKE ID Type. ## 1: IPv4_ADDR, 2: FQDN ## 3: USER_FQDN, 9: DER_ASN1_DN ## 11: Key ID ################################################## #SET NVIKEIDTYPE 11 ################################################## # ## IKE Xchg Mode. ## 1: Aggressive, 2: Identity Protect. ################################################## #SET NVIKEXCHGMODE 2 ################################################## # ## IKE DH Group. ################################################## #SET NVIKEDHGRP 2 ################################################## # ## IKE Encryption Algo. ## 1: AES-128, 2: 3DES ## 3: DEs 4: AEs-192 ## 5: AES-256 0: Any ################################################## #SET NVIKEP1ENCALG 0 ################################################## # ## IKE Auth algo. ## 0: Any, 1: MD5 ## 2: sHA-1 ################################################## #SET NVIKEP1AUTHALG 0 ################################################## # ## IKE Config Mode. ## 0: Enabled, 1: Disabled. ################################################## #SET NVIKECONFIGMODE 0 ################################################## # ## IPsec PFS DH group. ################################################## #SET NVPFSDHGRP 2 ################################################## # ## IPsec Encryption Algo. ## 1: AES-128, 2: 3DES ## 3: DEs 4: AEs-192 ## 5: AES-256 6: None ## 0: Any ################################################## #SET NVIKEP2ENCALG 0 ################################################## # ## IPsec Authentication Algo. ## 0: Any, 1: MD5 ## 2: sHA-1 ################################################## #SET NVIKEP2AUTHALG 0 ################################################## # ## Protected Network. ################################################## ### SET NVIPSECSUBNET 0.0.0.0/0, 0.0.0.0/0 ################################################## # ## IKE Over TCP. ## 0: Never, 1: Auto ## 2: Always ################################################## #SET NVIKEOVERTCP 0 ################################################## # ## Craft access ## 0: Enabled, 1: only view option is available? ################################################## #SET PROCSTAT 0 ################################################## # ## VPN craft access ## 0: disabled, 1: view only ## 2: View and edit. ################################################## #SET VPNPROC 2 ################################################## # ## Call Server address ################################################## #SET MCIPADD 192.168.42.1################################################## # ## craft access code ################################################## #SET PROCPSWD 27238################################################## # ## VPN craft access code ################################################## ## END

February 28, 2013July 9, 2014IP Office

How Do I Set Up Voicemail Pro Failover on Server Edition

Setting up voicemail failover on a Linux based Voicemail Pro, is pretty straightforward. Just like setting up an SCN trunk, pointing to each IPO, you need to point the Voicemail Pro’s to each other. With Server Edition, we only have 2 options for Voicemail Pro, Centralized and Backup. If an all IP500 SCN, we could also do Distributed. So, lets configure a Server Edition Failover.

In the examples below, Our Primary Server is 10.129.131.22 and our Secondary Server is 10.129.131.23.

First, I like to log into the Backup voicemail.