Voiceware
Aaron Bailey
Josh Roberson (Unlicensed)
Below is some basic info about Voiceware and setup. Page links to the left will provide more information about each section and feature of Voiceware.
Information on exporting Voiceware data for import to ConnectWare can now be found under the Voiceware Upgrade section below.
Current Version: 4.3.0.6
Release Date: 6/13/2024
New features: Added ability to change GUI password from command line.
Changes to CentOS in June 2024 might cause problems preforming a Voiceware upgrade. See the upgrade section below for more information.
4.3.0
Bug Fixes
Make voicemail for non-guest rooms work again (VW-309)
Fixed issue with PnP Provisioning not being reliable
If you assign a DID to an extension, and then subsequently reassign that extension as a DID, don't make the originally assigned DID vanish. (VW-311)
The original DID will now appear in RED on the extensions page and have "INVALID" as its searchable destination
Bring back the "Alert-Info: Internal" header for internal calls to admin extensions (VW-292)
Fixed an issue causing "DELETE ALL" on guests page to throw a PostgreSQL error. (VW-306)
Fixed an issue that seemed to specifically effect analog phones that came pre-programmed with XX**XXXXXX style speed dials, to fail to call those speed dials forcing a manual reprogram of those phones. (VW-307)
Fixed an issue where having certain characters in the hotel name field prevented hosted backups from working properly by preventing those characters from being used. (VW-267)
Fixed issue where Polycom firmware link wasn't showing up as a link. (VW-287)
Fixed issue where previewing an IVR from add extension selection screen caused multiple unwarranted "are you sure you want to leave" dialogs. (VW-326)
Fixed issue where French was causing malfunctions in the configurator due to "'" characters being in some words during translation causing js errors. (VW-328)
Fixed issue where all available languages were not available in settings tab for default system language (VW-349)
Fixed issue where incoming calls were not properly set to the system language selected on the settings tab for default system language (VW-350)
Fixed an issue that would cause email server configuration page to not save password if you clicked the show password box, unless you clicked that box again before saving. (VW-352)
Fixed an issue that would cause the Cancel button to not work unless the domain field was filled out on the email server configuration page (VW-351)
Fixed a display issue causing queue strategies not to show sometimes on the queue overview and details pages (VW-332/VW-333)
Fixed an issue that can cause speed dials for Polycom phones to become corrupted and point to '0' if the type isn't a "real" extensions (user,room). (VW-364)
Fixed a long standing issue preventing distinguishing between incoming external calls that use trunk cards on series2 cabinets. This also solves some cases where trunks feeding into series2 cabinets do not get ringback (VW-365)
Fix wakeup notify delivery notification to PMS via PSIP-PMS
Fix the ability to use external certificates by placing them in /data/asgi/ssl/
Slow down email delivery to prevent hounding relays with issues; also fixes bounce amplification
Reduce the amount of call loops caused by call groups/queues with forwarded devices by ignoring the forwarding from call groups or queues. (VW-340)
Fixed Polycom Guest Operations menu for SoundPoint IP phones (and views for VVX)
Fixed issue when recreating spans after upgrade to 4.x from older version (VW-362)
Reload res_parking.so after Parking Settings update (VW-370)
Fix manual uploading of phone templates (VW-212)
Fix issue where nightly automated audit might not send if the hotel name has certain characters in it (VW-376)
Fix an issue where sessions may not time out properly if browser tab or window is backgrounded/minimized. (VW-377)
No longer fail to match ACLs if customer copy+paste IP address with SPACES (VW-382)
Fix configuration and installation of DHCP for Express 4.x
Fix 500 when ACL script tries to log to syslog in some situations
Fix backup/restore lock files not being removed if the previous attempt failed unsuccessfully
SIP/PnP Provisioning works with SNOM phones again
VW-400 - Bugfix: Handle CID properly in edge case
Polyonymous no longer errors when it detects 802.1Q (VLAN) packets.
Calls are now billed and rated properly when forwarding (forwarder gets billed appropriately) (VW-410)
Enhancements:
Unified login page for both browser console and configurator (VW-313)
Should be an intuitive change, but if not, reach out.
Optimized Cleanup scripting that was sometimes taking lots of resources on newly upgraded systems due to the amount of CELs amassed when CEL cleanup was broken (Versions < 4.1.9) (VW-322)
Prevent non-CID characters from being entered in the default CID Number field on settings page. (VW-305)
Allow outbound rates to be editable by clicking their names, like most other items in tables are (VW-242)
Created a button that toggles all check boxes for ACLs (VW-237)
Removed Polycom specific wording from reboot alert. (VW-286)
Ability to download phone configs directly form endpoints page (replaces "Cfgd?" field) (VW-238)
Adjusted cookie warning banner to be less intrusive, still compliant with GDPR/Cookie privacy notice requirements. (VW-345)
Improved login and settings page (among others) load speed for configurator. (VW-263)
Apache enhanced security
Stop DAHDi from constantly reloading when a non-Lite system has spans. (found in-dev)
psip-pms page now has drop-downs for detected serial ports rather than a text field (VW-366)
prevents typos for serial devices
prevents accidentally using same port
Disable attended xfer keycode in asterisk that was accidentally getting triggered by elevator programming (VW-361)
Fail2ban is now restarted during 'voiceware restart all' or 'voiceware restart fail2ban' (VW-354)
By default email is now relayed reliably through Amazon SES so even if the customer does not have a local mail server, emails should reach the end party.
'voiceware' command now has proper help when called without arguments
'voiceware upgrade' now require a completed pref-light test to ensure system has sufficient resources
'voiceware upgrade' will now warn if multiple users are logged in so multiple users don't step on each other
'voiceware' command now locks and will not allow to users to perform conflicting actions
'voiceware status' command now reports the derived health of the Voiceware containers
'voiceware install' and 'voiceware upgrade' now uses 'dnf' instead of 'yum' for cleaner output
System upgrade "pref-light checks" no longer throws benign (ignored) errors
'voiceware fetch' now fetches the system upgrades as well as the Voiceware ones
DLR Data will now be sent to the browser console as well as the notification email, if configured. (VW-233)
Added "Set All Routes to use System Default Trunk" button on Outbound Routing page (VW-18)
Attempt to fix call loops caused by forwarding.
Note: this won't fix any loops caused by group fail overs, ONLY forwarding situations.
'voiceware sertest' handles stopping/starting of psip before/after doing the serial test.
'voiceware sertest' now uses the new sertest application written from the ground up
‘stunclient’ daemon can automatically update the external IP address on the system every 30-90 seconds
New Features:
Self-Service Password Reset for bc/configurator accounts.
Simply enter the username and click "Forgot Password?"
If user has an email on file an email will be sent.
Currently no password requirements during password reset to match current password schemes. Gross but we want this to be a call DECREASE not an increase because that they can no longer use 'FD1' for front desk passwords, so we'll not tighten the security here /just yet/.
Allow SSML in the TTS generation screen. If valid SSML is put into the text box, then it will be used for TTS instead of interpreting it as plain text. (VW-119)
Allow services on monitor page to be restarted from the GUI (VW-295/324/325)
Voicemail Transcription via email and/or configurator voicemail GUI (VW-15)
Will need some information added to the manual for this.
Can take UP TO 5 minutes to process, but usually less. Done via Amazon Transcribe
We check in 15s intervals for a max of 5 minutes before timing out
Error codes follow HTTP protocol codes with a subcode (.1/.2) and an epoch stamp for support log checking
Example: "Transcription Unavailable. CS:408.1.14843865334" (this means the voicemail was too long. The max length is unknown, but probably longer than voicemail can handle)
Add 'LMT' support to psip-pms and Voiceware. This will allow PMS' that support passing LMT to set credit limit on guests. (VW-291)
There are two special limit values:
999.99 will set the guests credit type to 'Open Credit'
0.00 (or empty) will set the guests credit type to 'No Credit'
All other values will set the guests credit type to 'Limited Credit' with the value passed in by the PMS.
Default masks are set in the application, but also in the packaged psip-pms.cfg. They are:
pmslmttag=3 MASK_LITERAL LMT
pmslmtroom=5 MASK_INT_LJUST EEEEE
pmslmtamt 7 MASK_STR_LJUST $DDD.CC
The dollar sign does not have to be there, but if it is, we account for it.
'voiceware diag' command performs basic diagnostics of your voiceware system and reports issues
'voiceware top' command will report realtime resource usage of voiceware containers
Directable Zero out of voicemail for users/standalone voicemail and guests. (VW-47)
There are two settings
Users/Standalone VM
Guests
Psip PMS USB Device (fixed in version 4.3.0.1)
A typo in the code causes the ttyUSB0 device not to open or be usable by Psip. This results in log file spamming.
To Fix
Setup the USB interface and select the correct protocol via the GUI.
When the interface restarts and you get the spam messages open an SSH session and edit the file
/data/psip/configs/psip-pms.conf
as root.Edit about line 15 from
device=/dev/ttyUSB0
todevice=/dev/serial/ttyUSB0
.Save and quit.
Issue the command
voiceware restart psip
.
After completeing these steps Psip shold function as normal. DO NOT restart the interface from the GUI as the wrong value will be written into the file again and the logs will start to be filled with garbage data.
4.3.0.1
Bug Fixes
Corrected an issue where selecting a USB to serial device caused Psip to fail to load correctly and flood the logs with error messages.
This only applied for a Serial to USB device. Serial connections were, and remain functional.
4.3.0.2
Bug Fixes
Fix for SMTP use port 587 instead of port 25 which is usually blocked by ISPs and hosting providers
Fix for DAHDI not starting if drivers not related to DAHDI do not build
New Features
Reduce verbosity of applying system changes to increase visibility of errors.
4.3.0.3
Bug Fixes
Fixed a rare issue with Queues that would cause the system to crash.
Fixed an issue with an underlining service that would cause high system load in some cases.
Fixed an issue where if a DID was added to the Dial Plan not as a DID type the settings page would not load.
Fixed an issue where if setting up a TCP/IP PMS connection the user would receive warnings about not using the same serial port for both the PBX and CA interfaces.
Fixed an issues for CDR viewing.
If an agent is part of more than one queue and weights for the queue are in use then a call to either queue would cause the system to crash.
Known Issues
UTC offset might not set correctly if the phones were provisioned from Voiceware. This is an underlining PHP issue. Refer to the Endpoint matrix to see if setting DST is available for a phone.
4.3.0.4
Bug Fixes:
Fixed an Asterisk crash after adding a new span.
Corrected an issue where Voiceware was not sending “pager” emails (for users).
Corrected an issue where the time zone offset was not working correctly.
Piaproxy health checks were not always working, corrected this.
Emails envelopes not being properly manipulated when using customer supplied email relay.
Disable sending phone refresh signals on configuration update.
New Features:
Added support for the Phonesuite Gateways (PSG24).
Updated MAC address identification for SIP phones and gateways (latest releases of phones and gateways should be correctly labeled in the Endpoints page).
Added Group code support.
4.3.0.5
Bug Fixes:
Added host entry for shortname to prevent dns lookups.
Changed default built in email server port from 25 to 587. This should help systems send email notifications.
New Features:
Added a new command “voiceware verify”. It looks at the current install and checks for any errors. This command might be slow to run but could help should a system not be working well.
Added device descriptions to “sip show peers” asterisk CLI command.
Added IP address and version number to console login screen.
A bug in Voiceware that would cause Voiceware to fail to start processes (psip in observed cases) and would not allow you to restart Voiceware. This bug has been fixed. This bug only effected version 4.0.3.5.
Details
When attempting to run the Voiceware status command (or other Voiceware commands) the below would display.
A lock file already exists.
This means a process is already running
Check with other technicians and/or support personnel
Fix
Run the command rm /tmp/.voiceware*
to remove the lock file and proceed. To permanently fix run the command yum install voiceware-osconfig
and answer yes when asked if you want to download the files. No restart or other action is required.
4.3.0.6
Bug Fixes:
Fixed the Voiceware command stuck lock file.
Fixed an issue where adding a custom speed dial with extension 0 would result in an error and the entry not saving the first time (but it would save if added again).
New Features
Added ring down settings to Endpoints.
Added polarity setting to Endpoints.
Enhancements:
Allow device secrets up to 255 characters.
Post-call record keeping optimizations (fix for systems crashing after paging a very large page group)
Updated the engine that TTS uses which will improve the quality of TTS sounds (existing sound files will be unaffected).
Sync pager 'from' and 'subject' lines with the email counterpart.
Removed the “Queue Callback” option from Queues (it was not functional).
Added the ability to change the GUI password from the command line.
Known Issues:
Turning on or off MWI from a host using a Proxy system does not work. MWI can be turned on manually via the A11 command only.
General Info
Default IP: https://192.168.1.222
Default Login: “installer” “Protect6607”
Support: 1-800-245-9933 or support@phonesuite.com
Calling to emergency numbers is something that all Phonesuite systems must do. This must be tested during installation and any time trunks are changed.
A video covering both Kari's Law and the Ray Baum's Act can be found here: Voiceware Compliance with Kari's Law & Ray Baum's Act.
Kari's Law
This law requires that all PBX systems allow both 9-1-1 and 9-9-1-1 to be dialed from any phone. It also requires a notification of a “location likely to be staffed”, in a hotel this is likely the front desk.
Voiceware is preprogrammed with 911, 8911 and 9911 in the outbound routes.
Make sure to set the default outbound route trunk in the Settings page and test 911 before leaving the site.
Ray Baums Act
This act in part reads:
(Sec. 506) The FCC must conclude a proceeding to consider adopting rules to ensure that dispatchable location is conveyed with 9-1-1 calls, including calls from multi-line telephone systems, regardless of the technological platform used. "Dispatchable location" means the street address of the calling party and additional information necessary to adequately identify the location of the calling party.
See https://www.congress.gov/bill/115th-congress/house-bill/4986.
Phonesuite had decided that the term "Dispatchable location" means the exact room number, the most exact location we are capable of providing. This is done in Voiceware by adding an emergency CNAM to either the room or device. Note that devices might need this if they are remote devices not at the main building at the physical address.
There is a 60 character limit on dispatchable location, this is imposed by the carriers.
Emergency Alerts
When a phone makes a call to 911 an alert box is displayed on the Browser Console. This warning displays the room number, name, and call length. As seen below it will display the room name of non-guest rooms, such as the Lobby in this example.
In addition there is a red banner at the top of the Browser Console that will remain as long as the call is in progress and can not be dismissed.
SIP Trunk Setup
SIP trunks must always be setup with 99 channels. Even if the provider only allows 10 channels for example you still setup the Voiceware side for 99 channels. Doing this ensures that Voiceware will not block any calls to the SIP trunk provider and places the burden of routing calls on the SIP trunk provider. Most SIP trunk providers should route a 911 call even if it exceeds the channel limit for the account.
Ray Baum’s Act requires that the dispatchable location (i.e. room number or location of phone) be sent to the 911 dispatch center on all 911 calls made from PBX systems. To comply with this requirement without needing a DID for each guest room a new feature has been added to Voiceware. This feature allows the dispatchable location (i.e. Room 101) to be dynamically passed when an emergency call is made, thus eliminating the need for a unique DID for every guest room and admin station.
There are currently two providers supported by Voiceware for Enhanced Dynamic Location Routing (hereafter referred to as "DLR"), http://Bandwidth.com and http://Bulkvs.com . This document will walk you through the provisioning of DLR entries for each service and the required information within Voiceware.
Process & Feature Overview:
There are four components to the enhanced DLR feature of Voiceware 4.2. They are:
An account with an approved SIP trunk vendor capable of passing the dispatchable location dynamically. Note that a compatible provider need only be used for emergency calls. Thus an existing provider can continue to be used for all normal inbound and outbound calling.
Dispatchable location settings in the Settings page. These differ depending on what SIP trunk provider you are using.
Dispatchable location information for Devices and Rooms. Each room and device entry must have a dispatchable location added to it. This is a simple line of text (60 characters max) that describes where the phone is (i.e. Room 101, back office 204, etc). This information must be added to the system and should be approved by the phone system owner (i.e. the hotel GM).
Pool of emergency call back numbers. Part of the emergency call requirements is that the 911 operator must be able to call back a number and have it ring directly to the phone that dialed 911. To comply with this requirement a small pool of DIDs (5-10) should be purchased on the compatible SIP trunk provider, not the existing provider.
Below are the setup instructions starting with the two approved SIP trunk vendors, note you only need to use one vender. Next is the Voiceware setup instructions and testing process. Lastly is a list of approved abbreviations for use in the Dispatchable location field and a SIP trace of a successful call.
Bulkvs.com
This guide assumes you have an account already set up with Bulk Solutions, and have e911 enabled on the account. Setting up the account and enabling the e911 service is outside the scope of this document. It is also assumed that you have purchased DIDs for the emergency call back pool, this is also outside the scope of this document.
Log in to your portal account at bulkvs.com .
On the left-hand side, click on “E911”.
Select “Add Endpoint”.
On the Add Endpoint screen, Fill out the required information. The Telephone Number is the TN you are setting up for 911 (ex. 13035551212).
The Subscriber Name should be the Site Name (ex. Hill Valley Mariott).
The Street Address will be the actual street address of the property (ex. 123 Maple St NW)
Enter “ng911” without quotes into the “Suite/Floor/Other” Field.
Then enter the City, State, and Zip code of the property in the appropriate fields.
DO NOT ENTER ANYTHING IN THE SMS NOTIFY FIELD.
Click “Validate & Provision”. Wait for the system to give a result. Correct any errors if present and repeat this step. If there are none, repeat steps 3-5 for each TN the site has (i.e. each DID in the emergency call back pool).
Go to the “Voiceware Provisioning” section and begin there.
Bandwidth.com
This guide assumes that there already exists a DLR account set up with Bandwidth.com; setting up such an account with Bandwidth is outside the scope of this document. Bandwidth is by far the more complex option currently and if you need help getting a special DLR account you much reach out to Bandwidth directly, Phonesuite will not be able to offer any help with the Bandwidth account.
Log into the Bandwidth portal and select (if not the primary) the account activated for DLR.
Click on the “Emergency” link at the top of the page.
Click on “+ ADD” under the “GEOLOCATIONS” section.
Enter the information for the site, using the site code as the “Geolocation ID”. Leave Address Line 2 blank. This is the CIVIC address or base address for the site.
Click “ADD GEOLOCATION” and wait for it to save. Correct any errors if any are detected and repeat this step, otherwise go on.
Click “ENDPOINTS”.
Click “+ ADD” under the ENDPOINTS section.
The Endpoint ID will be the full TN including the 1. (ex. 13035551212)
The Caller Name should be the Site Name (ie, Hill Valley Mariott)
The Callback Number will be the TN minus the 1 (ex. 3035551212).
Click the “SEARCH GEOLOCATIONS” button, and type in the site code and click “SEARCH”.
Next to the Geolocation ID click “SELECT”.
Click “ADD ENDPOINT” and wait for it to save. Correct any errors if there are any, and repeat this step. Otherwise, repeat steps 7-13 for each TN the site has.
Once this step is completed, Make a note of the Account ID (it’s in brackets [] after the account name on the Overview section, and the Site Code. You will need these to provision Voiceware.
Go to the “Voiceware Provisioning” section and begin there.
Voiceware Provisioning
NOTE: you must finish the SIP provider section above before proceeding with the Voiceware setup.
Dispatchable location settings in the Settings page
Log into the Voiceware system as Installer (or another admin user).
In the “System” section, click on “Settings”
Scroll down to “Emergency Calling Settings”
Option 34: Select an extension to notify in the event of an emergency call. This will usually be the front desk or operator extension.
Option 35: Enter an Email address to send emergency notifications to. These emails will tell the recipient the number dialed, and the extension that dialed them, along with the date and time the call was placed. If you leave this field blank, no such emails will be generated.
Option 36: Ray Baum’s Act requires the ability to call back the emergency caller for an allotted time after a disconnect of the emergency call. By default, the TN selected by the system will be reserved for 60 minutes as a direct-dial number to reach the calling device. This can be adjusted as needed, but 60 minutes or greater is recommended.
Option 37:
For http://Bandwidth.com users, this will be the following:
Geolocation: <https://emergency.bandwidth.com/locations/{account_number}/{location_id}?loc={device_location}>
For Bulk Solutions (http://bulkvs.com ) users, this will be the following:
X-ng911: {device_location}
Option 38: BANDWIDTH ONLY: Enter your Bandwidth Account Number.
Option 39: BANDWIDTH ONLY: Enter the Site Code (or other Geolocation ID if site code not used (rare)).
Option 40: If you’re using Bandwidth, check the box for URLEncode. If you’re using Bulk Solutions, make sure the box is unchecked.
Click “Save all Changes” and wait for the page to refresh.
Dispatchable location information for Devices
Click on “Devices” at the top of the page.
Click on the name of the SIP device to add location data to.
In the “Dispatchable Location” Field, enter the pertinent location data. (ex. FLR 4 RM 410, for Floor 4 Room 410 – see appendix A for preferred abbreviations).
Click “Save”.
Repeat steps 11-15 for each SIP device that is not a trunk or proxy device.
Dispatchable location information for Rooms
Click “Rooms” at the top of the page.
Click on the Room Name to add location data to (location data on devices assigned to rooms is ignored in favor of the room entry.)
In the “Dispatchable Location” Field, enter the pertinent location data. (ex. FLR 4 RM 410 for Floor 4 Room 410 – see appendix A for preferred abbreviations).
Click “Save”.
Repeat steps 17-19 for each room you need to add location data to.
Note: Dispatchable location data can be added in a CSV bulk room upload. See the Rooms page help page for more information.
Pool of emergency call back numbers
Click “Dial Plan”
For any TN such as the main number or reservations number, you will want to block the 911 system from re-routing that TN when an emergency call is placed:
Click the TN (DID) you wish to block from the 911 system
Check the box labeled “Block Emergency Use”
Click “Save”.
If you purchased pool DIDs for emergency use (should have 5 at a minimum) you will add these to an emergency pool. Click “DID Pooling” in the “Extensions” section on the left side of the screen.
Click “Add Pool”
Name will be an identifier such as “Emergency TNs” or “DLR Numbers” or something to identify this pool of TNs.
Check the box “Emergency Callback Pool”
Select a Destination for the pool. This destination is where incoming calls to these TNs will ring when they are not being used for emergency purposes. Usually, the front desk or an IVR that plays a “not in service” message.
Click “Save”
Click “Dial Plan”
Click “Add Extension”
Select a number that would not be easily dialed by accident, such as **5959.
Extension Type should be “DID Pool”
You can add the first DID to the pool by adding it with the DID assign/add selection if you wish. Otherwise leave it as “None”.
Destination will be the pool you just created in step 24 above.
Click “Add.
If you have not yet added the emergency use TNs to the system:
Click “Add Extension”
Enter the TN as the Number/Pattern
Extension Type will be “DID”
Destination will be the pool you just created in step 24 above.
Click “Save”
Repeat this process for each TN you need to add.
If you already have the emergency use TNs added to the system, but not pointed at the pool:
Click on the TN to assign to the pool.
In the “Edit Extension” page, change the Destination to the pool you just created.
Click “Save”
Repeat this process for each TN you need to reassign.
Testing
You are now ready to place a test call from one of the configured devices. Make sure you have your routes set up properly (including a route for 933 as an emergency route) and trunk devices for the appropriate carrier set up. Dial 933 from a device on the system and you should hear back both the number the system selected for callback (one of the emergency use TNs) along with the address and location specific details.
Note on how TNs are selected:
First, the system will utilize a TN assigned to the user/room/guest/etc. who made the call, if available. If not, the system will check for a previously used TN in the expiry period for the device making the call and re-use that TN. If a TN is still not found, the system will then choose a TN from any DID pool(s) marked for emergency use. If there are none available (already used and within timeout period or none defined) the system will choose a TN from the system that isn’t marked with the “Block Emergency Use” flag. The TN that is chosen in any instance will be directly linked with the device making the emergency call for the expiry period and not usable by anyone else on the system until the expiry period is over. This is to ensure a direct call-back from the emergency call taker.
Support: 1-800-245-9933 or support@phonesuite.com
Return Merchandise Authorization (RMA) is given for items under warranty only. When there is a problem with equipment, a tech will try to assist you via phone to resolve the issue. They may advance a “like” item if your system will not function without the part and we need to test the equipment.
DOA WARRANTY REPLACEMENT: If the item is “dead on arrival” (within 30 days of shipping), we will follow the normal warranty replacement procedure (below) but will also pay overnight shipping costs. WARRANTY REPLACEMENT: If the item is covered under the two-year warranty, the item will be replaced at our cost including ground shipping (you can pay the difference if you’d like shipping other than ground). Equipment serial numbers must be verified and you must be able to tell us what hotel the part came from to verify registration. Warranties are void if the system has not been registered. New or used equipment is sent at our discretion. An invoice for the full amount of equipment is created at the time advance equipment is sent. If we receive the old equipment within 14 days and it is determined to be covered by warranty (no lightning, spills, misuse) the invoice will be voided. If the return equipment is not received within 14 days the invoice is sent to you and payable immediately. We will not accept return equipment back after a total of 30 days and you must pay full price for the equipment that was advanced plus all shipping charges.
Staying up to date
We produce, on a regular cycle, updates that correct or expand functionality in Voiceware. As part of the update process, the OS packages (currently based on CentOS 7) are updated as well. Regularly checking for upgrades and applying both Voiceware and OS upgrades is best practice.
Phonesuite IPs
Phonesuite Offices: 23.24.133.24/29
Huntsville Offices: 75.6.52.128/29
Netlink Offices: 38.65.54.14/32
Dallas Offices: 107.134.201.104/29
Required Ports
Below is the list of required ports and their uses.
Port | Protocol | Application | Use |
22 | TCP | OpenSSH | Remote administration via SSH |
69 | UDP | TFTPServer | Phone configuration download via TFTP |
80 | TCP | Apache/HTTP | Phone configuration, letsencrypt verification, and PSIP XML-RPC |
123 | UDP | NTP | Provide time information to phones and other servers/services |
443 | TCP | sslh | SSH/HTTPS Proxy, SSH Administrator/ HTTPS Web UI |
5060 | UDP | Asterisk/SIP | Standard UDP SIP port for phone/carrier communication |
5060 | TCP | Asterisk/SIP | Standard TCP SIP port for phone/carrier communication |
5061 | TCP | Asterisk/SIPS | Standard SIP TLS port for phone/carrier communication |
8080 | TCP | PSIP/XML | PSIP-PMS XML RPC Server for Voiceware communication |
10000-20000 | UDP | Asterisk/RTP | RTP Ports for phone/Carrier audio |
Outbound Connections
Voiceware will make outbound connections to several different locations and services.
License server: http://license.voiceware.com TCP ports 80 or 443 outbound only
Voiceware Repo: http://registry.voiceware.com TCP port 443 outbound only
Voiceware Management: salt.mgmt.voiceware.com TCP port 4506 outbound only
Cloud Backups, TTS: See https://docs.aws.amazon.com/general/latest/gr/aws-ip-ranges.html outbound only
If using remote CDRs:
CDR Server: asteria1.voiceware.com TCP port 443 outbound only
The systems will also need to be able to reach the distribution servers for CentOS, Debian, and Ubuntu. All Outbound only.
NTP:
server http://0.centos.pool.ntp.org
server 1.centos.pool.ntp.org
server http://2.centos.pool.ntp.org
server 3.centos.pool.ntp.org
Voiceware Firewall
The Voiceware appliance comes with a firewall pre-configured. It uses firewalld to manage the iptables in-kernel firewall.
By default it allows connections only to the necessary ports: 22, 69, 80, 123, 443, 5060, 5061, and 10000-20000.
Because psip-pms does not have an authn or authz method, it is not enabled by default. It is recommend you create a separate zone for allowing communication into the psip-pms ports (8080 by default).
Fail2ban
Fail2ban comes pre-configured on the default installation of Voiceware. It scans the logs and blocks offenders based on repeated authentication attempts.
fail2ban does not block IPs in the RFC1819 range: 10/8, 172.16/12, 192.168/16
Sometimes in a hosted environment it’s wise to whitelist the sites public IP so that it never gets blocked by Fail2Ban. To do this:
Log into the shell via SSH (recall that port 443 can be used when port 22 is unavailable).
Edit the file using the command
sudo vi /etc/fail2ban/jail.d/00-ignoreip.conf
At the top of the file is a line called “ignoreip” with a listing of other IP addresses.
Important! Do not remove any of the existing IP addresses.
Add the sites public IP either directly or with a slash subnet notation at the end of the existing IP addresses (using a space as a separator).
Save the file by entering
:wq
Issue the commandsudo service fail2ban restart
.
To whitelist IPs for the PMS TCP/IP setup use the below commands:firewall-cmd --permanent --new-zone=pms
firewall-cmd --permanent --zone=pms --add-port xxxx/tcp
firewall-cmd --permanent --zone=pms --add-source=x.x.x.x
firewall-cmd --reload
To test you can also run these commands:firewall-cmd --get-active-zones
firewall-cmd --list-all
firewall-cmd --list-all-zones
Voiceware 1-3 use the commands:
iptables -I INPUT -p tcp --dport xxxx -j ACCEPT
iptables -I trusted -s x.x.x.x -j ACCEPT
service iptables save
or/etc/init.d/iptables save
10k-20k/UDP (RTP/Audio) It is generally recommended to create a separate network for the Voiceware system and all IP phones. It is not recommended that the Voiceware server be on the guest LAN. If this is unavoidable it is recommended that the Voiceware ACL list (accessed form the settings page) be used to block access to all non-hotel employees.
Opening port 22 is not normally needed and is a frequent point of attack. Never open port 22 without a strong Access Control List (ACL) in place.
SSH access is available on port 443.
SIP phones can auto configure to the system via TFTP if they are on the same LAN, if the phones are remote, or the system is hosted, then port 80 (HTTP) must be used (http://<ip>/p). See the SIP Auto Provision section. Auto Provisioning
If you open port 5060 but not the RTP range (10,000 – 20,000) SIP calls will setup but will have no audio.
Voiceware is very flexible and can be setup any number of ways. Below is a set of common best practices when building the system. The goal with these is to create a system that is easy to setup and troubleshoot later. We feel strongly that creating and using standards between systems will lead to higher quality and a quicker installation.
Users
Usernames should always be the position name (i.e. FD1, FD2, GM, AGM, Sales1, Sales2) and never a users actual name (i.e. Scott).
Create a standard list of usernames and always pull from this list, adding numbers when needed. FD, GM, AGM, Sales, Admin, Res, Kitchen, RS (for room service), Porter.
Rooms
Always use the user or room extension as the SIP Device name.
Room names should always be “RM” + their room number (i.e. Rm 201). This should not be changed for different room types (i.e. suites).
Name non-guest rooms with their extension and function or location. For example, “302 Cordless” or “450 Aspen Room”. This will aid the Brower Console users as this name is displayed there.
System Features
Label call groups with the department or function (Operator, Front Desk, Sales, Reservations). Never use shorthand or codes (i.e. FD or 34V)
Group like extensions into ranges, for example the front desk locations should be 501, 502, & 503. Not 501, 542, & 552.
If using lots of call groups, queues, or conference bridges agree on a reserved extension range for each system feature (i.e. 84X for Queues, 85X for conference bridges)
Conference bridges, if one is assigned to each meeting room, can also be given extensions of 4+room number (i.e. 4205, 4213) or something similar that does not overlap with guest room numbers.
Always setup your local network IP range the same and always assign the same devices the same static IP address (i.e. always assign Voiceware 192.168.1.91, gateway 1 192.168.1.92, etc)
IVRs
IVRs should have extensions in the 8XX range unless this overlaps with room numbers. In that case the IVRs should all be given 4-digit extensions in the 8XXX range.
Build an IVR called “DID not assigned” that simply routes calls to the operator call group. Then when adding DIDs to your system assign them all to this IVR. Doing that will make it obvious what DIDs have been intentionally assigned to the operator call group and what DIDs have not. This is a way to help manage a hotel with a block of 30 DIDs that only has 3-4 listed as their main number or assigned to users.
Name all IVRs with their function, avoid codes or shorthand.
Installing a New Cert
You must first create an FQDN for your site via a DNS provider. We recommend DNS Made Easy | Fast and Most Reliable Provider but there are many DNS providers.
Make yourself root
sudo -i
Issue the command:
voiceware ssl
Follow the on screen prompts. First it will ask you for an email address, this should be the default support email address your customers use. Next it will ask you to agree to the terms and conditions. Then ask if you would like to send data to Lets Encrypt to improve the product (say no). Lastly it will ask for the FQDN without the https data. So for example if my site is at https://julia.voiceware.com I would enter just “julia.voiceware.com”.
Renewal
Every 90 days the certificate must be renewed. The renewal process reaches out to the Lets Encrypt servers which in turn triggers an inbound connection from a Lets Encrypt verification server. This verification comes from a different IP and thus if the sites firewall is not setup to accept HTTPS and HTTP traffic from anywhere (a best practice) this verification will fail.
Renewal triggers an inbound connection from a different IP than the renewal process uses.
To trigger the renewal process manually issue the command on the CLI.
4.X docker exec -it webapps certbot renew
3.X /root/certbot-auto renew && service apache2 restart
Below is a list of IP ranges that must be whitelisted so the verification process works.
IP | Date Added |
13.58.0.0/15 | 11/28/2022 |
18.116.0.0/14 | 11/28/2022 |
18.159.0.0/16 | 11/28/2022 |
18.192.0.0/16 | 11/28/2022 |
18.220.0.0/14 | 11/28/2022 |
18.237.0.0/16 | 11/28/2022 |
23.178.112.0/24 | 11/28/2022 |
3.120.0.0/16 | 11/28/2022 |
3.128.0.0/15 | 11/28/2022 |
3.132.0.0/14 | 11/28/2022 |
3.136.0.0/13 | 11/28/2022 |
3.16.0.0/14 | 11/28/2022 |
3.20.0.0/14 | 11/28/2022 |
3.69.0.0/16 | 11/28/2022 |
3.74.0.0/16 | 11/28/2022 |
34.208.0.0/12 | 11/28/2022 |
35.80.0.0/12 | 11/28/2022 |
54.184.0.0/15 | 11/28/2022 |
54.200.0.0/15 | 11/28/2022 |
64.78.144.0/21 | 11/28/2022 |
66.133.108.0/22 | 11/28/2022 |
Fortinet Router IP Addition Script
config firewall address
edit "SSL 2022-B 1"
set subnet 13.58.0.0/15
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 2"
set subnet 18.116.0.0/14
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 3"
set subnet 18.159.0.0/16
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 4"
set subnet 18.192.0.0/16
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 5"
set subnet 18.220.0.0/14
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 6"
set subnet 18.237.0.0/16
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 7"
set subnet 23.178.112.0/24
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 8"
set subnet 3.120.0.0/16
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 9"
set subnet 3.128.0.0/15
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 10"
set subnet 3.132.0.0/14
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 11"
set subnet 3.136.0.0/13
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 12"
set subnet 3.16.0.0/14
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 13"
set subnet 3.20.0.0/14
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 14"
set subnet 3.69.0.0/16
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 15"
set subnet 3.74.0.0/16
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 16"
set subnet 34.208.0.0/12
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 17"
set subnet 35.80.0.0/12
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 18"
set subnet 52.36.0.0/14
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 19"
set subnet 54.184.0.0/15
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 20"
set subnet 54.200.0.0/15
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 21"
set subnet 64.78.144.0/21
set comment "2022 Group B 11/28/2022"
set color 21
next
edit "SSL 2022-B 22"
set subnet 66.133.108.0/22
set comment "2022 Group B 11/28/2022"
set color 21
next
end
config firewall addrgrp
edit "2022 SSL Group B"
set member "SSL 2022-B 1"
append member "SSL 2022-B 2"
append member "SSL 2022-B 3"
append member "SSL 2022-B 4"
append member "SSL 2022-B 5"
append member "SSL 2022-B 6"
append member "SSL 2022-B 7"
append member "SSL 2022-B 8"
append member "SSL 2022-B 9"
append member "SSL 2022-B 10"
append member "SSL 2022-B 11"
append member "SSL 2022-B 12"
append member "SSL 2022-B 13"
append member "SSL 2022-B 14"
append member "SSL 2022-B 15"
append member "SSL 2022-B 16"
append member "SSL 2022-B 17"
append member "SSL 2022-B 18"
append member "SSL 2022-B 19"
append member "SSL 2022-B 20"
append member "SSL 2022-B 21"
append member "SSL 2022-B 22"
set comment "2022 Group B 11/28/2022"
set color 21
end
2023 Case
This case includes a better CPU (i3-6100), more RAM (32GB) and a faster MSATA SSD (512GB).
2019-2023
Phonesuite will ship a newer 1U rack mount server case with all 4.1 orders where an expansion card is not required. These new cases are silver and have two USB 3.0 ports on the front that do not work (our current board does not have headers for them). This case does include two lights on the front showing the status of NIC 1 (LAN) and 2 (Series2 Span).
Device | Watts | Current | Volts | Frequency | Kwh / Day | PF |
Rack Mount Server | 12.5 | 195 | 119.65 | 60 | 0.29 | 0.529 |
Series2 | 21.21 | 186 | 119.50 | 60 | 0.53 | 0.964 |
Environment:
Operating Temperature : 0 ~ 60°C
Storage Temperature : -10 ~ 70°C
Voiceware has three access points, they are:
Configurator: the admin portal where most setup and work on the system will be performed. This GUI is what most of the manual covers.
Browser Console: A simple web GUI for front desk staff and other system operators. Used to setup wakeup calls and make other basic changes (i.e. calling restrictions, day/night mode, etc.)
SSH: This is the command line interface to the Linux operating system. It’s rarely used but some items in the How-To section refer to it. Phonesuite support can always be contacted to offer assistance in interfacing with the Linux OS if needed.
Voiceware: A hospitality specific IP based phone system. Basically, Voiceware is a Linux based computer application that manages all phone calls for a hotel.
PBX: Stands for Private Branch Exchange, general term used to refer to a system that allows calling without interfacing with the local telephone company. Voiceware is a hospitality specific PBX.
Trunk: A trunk is a connection to a telephone provider. This connection can come in the form of standard copper telephone lines, T1/E1/PRI digital lines, or SIP trunks.
PMS: Stands for Property Management System, this is the main core computer system in a hotel. This system manages all guest information and interfaces with other system as needed to send and receive guest related data. Voiceware often interfaces with a PMS to receive check in/out data and guest names, we also send to the PMS information about billable calls a guest makes.
SIP Phone: Refers to a type of phone that is a network device where calls are sent and received as network data. These phones are often found at admin stations because they provide many more features than analog phones.
Analog Phone: Most commonly found in guest rooms these simple phones often have no display and will not always have speed dial buttons. Analog phones require an additional piece of equipment to interface with an IP based PBX (like Voiceware) and SIP phones. Phonesuite offers the Series2, and 3rd parties offer gateways that fill this function.
Installation Disk – The installation disks are generated by a program called KIWI, currently maintained by the OpenSUSE Team. The installation is streamlined so that it ships with the most basic of components, but does include an onboard copy of voiceware-osconfig package with the repository information baked in. The system comes in ISO (hybrid) and TFTP media. The factory is currently installing via TFTP from a server located on the CMP/Phonesuite HQ network.
Operating System – The base of the operating system is CentOS 7. CentOS is a rolling release within major versions, meaning an earlier CentOS releases are compatible with later releases, meaning that upgrades are safer to perform. CentOS 7 is currently supported until June 30, 2024. The voiceware-osconfig package relabels the system as “Voiceware OS”. While this is not a fully separate distribution of CentOS, the labelling allows us to denote that it is being configured and maintained by Phonesuite/Voiceware scripts
OS Configuration – The main OS configuration is handled by a SaltStack recipe that is distributed by the voiceware-osconfig package stored on the Voiceware download mirror. The recipe is responsible for setting up:
System security (Users/Groups/sudo, firewall, keys and encryptions)
Packages (including upgrades)
Various system configurations required for basic use
DAHDI installation
SSH Access – The remote login differs slightly from the previous versions of Voiceware. Now instead of using a single user with a single password, we have moved to a more secure SSH Keys. The support user is now ‘support’ and the ‘phonesuite’ user is reserved for resellers. When in the field, once the reseller logs in the first time, a secure password is required to be set to continue. If using it in the PSD configuration, this user should be configured and locked before leaving the factory or shortly after
Daemon Control / Monit – System daemons are no longer controlled by monit and monit is no longer used. SystemD has now taken over most of what Monit did, start/stop/ensure processes stay running. SystemD is also the main process (init, pid 1) for CentOS/Voiceware OS. The systemd utility accepts the following actions: start stop restart status.
System Journal / Logging – The new system contains a unified logging system. The logs are initially written to an internal (in-memory) journal before being flushed to disk. The in-memory journal can be accessed with ‘journalctl’. This will display the ENTIRE journal that is in memory in ‘less’ and you can use the familiar ‘less’ keys to navigate. If the time/date you need is not available in the online journal, the journal is flushed to disk and stored as /var/log/messages, with the familiar rotation. To view only a specific process/services logs, you can use ‘journalctl -u <service name>’, or you can view a specific syslog tag by using ‘journalctl -t <tag>’.
Dispatcher – An installable daemon (voiceware-dispatcher) that handles communication between unpriviledged containerized software and the Operating System. This daemon reads a list of commands from the database and inserts the responses accordingly.
Logs: journalctl -u dispatcher
Docker - Docker is a set of products that uses OS-level virtualization to deliver software in packages called containers. Docker is used to manage the Voiceware Software itself. It controls the building of the software packages, deployment (via a download portal), starting, stopping, and maintenance.
Docker Storage – All user-accessible files are stored in the /data directory on the OS. These files may or may not be updated via automated processes within the containers.
Voiceware Containers - While it is possible to access the containers and modify files directly, this is highly discouraged as the changes may be lost on container restart and will be lost on upgrade. Instead any modifications needing to be made inside the container should be sent to engineering for review.
Postgresql – This container is the main database process. It stores all of it’s data in /data. There are no user-serviceable components. The alias ‘psql’ is configured to talk to this database for scripting needs.
Logs: journalctl -t database
Asterisk – This container is the main asterisk process. the alias ‘asterisk’ is configured to talk to asterisk within this container. There is no mechanism to change any configuration files for asterisk.
Logs: journalctl -t asterisk
PAI – PBX Abstraction Interface – The main messaging bus for all telephony components. This container has no serviceable parts
Logs: journalctl -t pai
PAIProxy – Websockets proxy for the PAI process. This is used by the Browser console to communicate with PAI. Container has no serviceable parts.
No Logs
PSIP – Phonesuite PMS Interface software, originally developed by Frank Melville. Data for this container is stored in /data/psip and should be configured via the web interface. This container has no serviceable parts.
Logs: /data/psip/logs/
Omegadial – Blast dialer used to handle emergency notifications. All of the process’ data comes from the database and is configured via the web interface. No serviceable parts inside.
Logs: journalctl -t omegadial
SIPNP (ships inside phpapps image) - PNP Responder for Vtech/Snom/Yealink SIP boot provisioning. No serviceable parts, but has extensive logging.
Logs: journalctl –t sipnp
TrunkTester – Trunk testing daemon. It is designed to dial out to a number and test audio by sending DTMF (key presses). Configuration comes from the web interface. No serviceable parts.
Logs: journalctl -t trunktester
Webapps (ships inside phpapps image) - Main web interface. Contains both configuration and Browser console components. This image also feeds licenses to all subcomponents, so must be running to enable. Downloads necessary templates and help files on start. No serviceable parts.
Logs: journalctl -t webapps
Polyonymous (ships inside phpapps image) - Phone Discovery Daemon – Listens to ARP requests on the network and attempts to derive make/model/software based on network events. No serviceable parts.
Logs: journalctl -t polyonymous
JAIS – Main call routing software. Handles all the nitty-gritty call routing details. No serviceable parts.
Logs: journalctl -t jais
Scheduler (ships inside phpapps container) - Main cron/scheduling daemon. Handles setting up the database and filesystem on initial start. There is also an unprivileged version of the dispatcher running here that enables cross-container communications.
TFTPServer – Contains the TFTP server and (shipped) firmware files. Firmware is copied to /data/tftpboot on container start. No serviceable parts.
Reporter – Contains the legacy Asteria queue reporting running on a very old version of ubuntu. Connections to this are proxied via the main Webapps container/web server. No serviceable parts.
Voiceware training is hosted online at Phonesuite Training. Email Sales@Phonesuite.com to request access to the training.
This sheet is intended to aid installers when gathering information from the hotel. This is not intended to be a full and complete list, just the most commonly needed information.
Below is a basic set of questions and pieces of information you need to have on hand before you attempt to configure and connect a PMS interface to Voiceware.
Below is an installation and setup check list this is intended to be used during an installation to ensure that no critical setup is missed. Unlike each section chapter the check list will not cover every item in each section but simply the things that need to be changed most.
Settings
Enter the hotel’s name in [Hotel Name, City, State] format
Enter the systems public IP address and local subnet
Enter the e-mail address that is notified of system events
Set up and test the PMS interface
Check in a test guest and check for proper name and dialing restriction
Set emergency call notification extension
Set wakeup call time in-between calls and total reattempts
Enable ACS’s after adding your own offices public IP address (Optional)
Users
Add hotel staff as Users, assign extensions and devices (if applicable) at the same time, list all hotel staff in the directory and remember if they there is no last name entered the staff member will be unsearchable from the directory
GM
Front Desk
Sales
Restaurant / Bar
Porter
Rooms
Use bulk add room to quickly add rooms and dial plan extensions
Add all non-guest rooms (conference rooms, break rooms, etc…) and set dialing permissions as appropriate
Check that a rooms message waiting lamp functions properly
Devices
Add any devices not entered during the users set up and insure that the following fields are set properly
Device name: Should be the extension number
Assign to: Select a user or room
Friendly Name: the users full name or the name of the room the device is in
Secret (for SIP devices)
Username: same as Device Name, should be the extension
SLA Group (if applicable)
Dial Plan
Add defaults to Outbound Routes
Setup digit stripping for failover outbound routes (if failover is being used)
Ensure that an extension has been created for each device
Assign an extension to each conference room
Add DID extension types for each inbound DID
Assign an extension to each Call Group and add members to it
Assign an extension to wakeup call setup
Call Group
Assign members to the operator Call Group
Set up any other needed Call Groups and assign members to them
Conferences
Name the conference room in a way that makes it easily identifiable
Enter a User and Admin pin if needed,
IVR
Map out existing IVR by calling it and going through the prompts or obtain a copy of the IVR map from the hotel staff
Add the IVR and name it based on its function
Add all steeps to the IVR and check once complete that there are no undefined steeps and that there are no steps that are hanging in limbo (steps that don’t have arrows pointing to them)
Sounds
Upload sound files and name them appropriately
Add the new sound files to Drop-In VM Messages or into a Queue as needed
Miscellaneous
Test several room phones for functionality
Check to make sure the BIOS clock is set correctly and that power on after power failure is set
Ensure that Emergency calls function properly, perform at least one test 911 call
Test speed dial buttons on existing analog phones to ensure functionality
Test ALL SIP phones for functionally
Train hotel staff and management on the proper use of the Browser Console
Have hotel staff test basic functionality (calling a room, transferring a call, checking in a guest, etc…) to make sure they know how to do it and don’t have any questions about it.
Test inbound and outbound calls
Confirm that hotel management has installer’s contact information
Thank the staff for their help and patients during the install and thank them for choosing Voiceware by Phonesuite
Voiceware Upgrade
This document describes how to install Voiceware 4.2 in various scenarios and on different hardware. Please read the entire section before beginning any installation or upgrade process.
Warning! DO NOT attempt to reimage a live system without having a temporary system in place. If this is done, and support is needed, that support is billable!
In June 2024 CentOS made changes which can cause issues with Voiceware upgrades.
4.3.0.X: No changes needed, the voiceware upgrade
command works without issue.
4.2.X or older: Run the command dnf install voiceware
first then the command voiceware upgrade
.
Install from ISO: Broken at this time. Issue the command curl https://get.voiceware.com/install | sh
to kick off the installation.
Sections
Creating a Bootable USB Key: A bootable USB key is used to install Voiceware on physical hardware and is a required step for the temporary system method of existing system upgrades.
Physical Hardware: Temporary Server Method: this section outlines the process for upgrading an existing site from a previous version of Voiceware (i.e. 1.X, 2.X, or 3.X) to version 4.2 using a temporary server to run the hotel while the existing hardware is being upgraded.
Physical Hardware: Hard Drive Replacement Method: here an upgrade on existing physical hardware is accomplished by swapping out the systems hard drive for one that has 4.2 installed on it already.
Voiceware 4.2 Installation: this section describes the installation process of Voiceware 4.2 from the command line interface.
VMware Workstation 16: Here a description of installing Voiceware 4.2 in a virtual environment is outlined. This is most often used for demo, testing, or training purposes.
Other Hosted Providers: There are many other hosted providers that exist. This section provides information on the type and power of hardware needed should a hosted provider other than AWS be used.
Testing & Troubleshooting: A short check list of the types of things that should be tested after the upgrade of any existing site and common fixes should the test fail.
Planning
Check resources (Documents, equipment, factory support if needed)
Recovery plan
Time & Schedule
Battle Plan (Checklists, process documents, etc)
Early Prep
Review database for duplicate outbound routes
Restore database to temp 4.2 system or in house test 4.2 system
Document
Span MAC addresses
Notes in setup
Counts of registered and unregistered devices (screenshots)
Go/NoGo
Use a check list
Don’t be afraid to abort and try again once issues have been addresses
Notification
Hotel staff
Any other resources such as office support staff or Phonesuite factory support
Pre-Cut
Setup
Required equipment
Extra equipment (optional)
Connections and methods to monitor in place
Backups and docs
Wakeup call list
Checked in rooms
Create and download a backup
Setup
Upgrade
Monitoring
Post Upgrade process
Update/confirm license
Set time zone in settings page, restart if needed
Rebuild spans
Confirm registration count
Internal call tests
Inbound call tests
Outbound call tests
Advise property staff of restoral to service
Check CLI for error messages
Check logs for error message
Email factory or other ‘standby’ staff of success or issues
Disaster Recovery
Voiceware can be installed on just about any hardware using a bootable USB key. An existing system (either SBC card or stand-alone 1U rack mount server) can also be “rekeyed” using this process. Note that this process DESTROYS all the data on the system.
Download Rufus from: Rufus - Create bootable USB drives the easy way
Download the current Voiceware ISO from:
Insert a BLANK USB key (min 2GB) into your system
Launch the Rufus tool
In the Device dropdown select the USB key
In the Boot section click the Select button and choose the Voiceware ISO downloaded above
Set Partition Scheme to MBR
Set the Target System to “BIOS or (UEFI-CSM)” (might be listed as “BIOS or UEFI”)
Click START
10. If asked what method to use, select DD and press OK
When finished the USB key can now be inserted into any system and when that system is rebooted it will begin loading from the USB key. Note that during boot up F7 or F10 might need to be pressed to select to boot from the USB key. If the system will not boot from the USB key adjustments to the BISO might need to be made, please contact Phonesuite for assistance in doing this. At this point begin at step 1 in the Voiceware 4.2 Installation section below.
This method assumes a live in the field system is being upgraded from a previous version of Voiceware (i.e. 1.X, 2.X, or 3.X) to version 4. Note that a system already on version 4 can simply be upgraded without the need for a complete reinstallation as outlined below.
Prepare: For this type of upgrade, you will need
A temporary 1U rack mount system loaded with the latest 4.2 installation
Contact Phonesuite for a temporary server if needed
A bootable USB key (see the section “Using a bootable USB key to install on physical hardware” page 3 for how to create a bootable USB key)
Two 10-15ft network cables
Long power cord or extension cord (depending on phone room power availability)
A laptop
A screen and USB keyboard.
Check the existing system for duplicate outbound routes, outbound rates, and extensions. Older versions of Voiceware allowed duplicates however a backup containing duplicates will not load into a 4.2 system.
At a high level the process here is to use the temporary server to run the hotel while the physical hardware is upgraded. Once done the hotel is put back on the existing hardware running Voiceware 4.2. This method minimizes downtime and allows the hotel to continue running on the temporary should there be a hardware failure during the upgrade process.
In office prep work (optional but recommended)
Before traveling to the customers site the below prep work should be completed, this will minimize onsite downtime.
Verify the system is licensed. Contact Phonesuite support to obtain a new license for the temp system as needed.
Verify the temp system is running the latest version of Voiceware.
Upload the customers database to ensure it will be successful and know how long it will take
WARNING: take care with registered SIP trunks. If the trunks register to the system in your office calls to the hotel might start ringing the temp PBX and not the hotel. To avoid this block port 5060 in the router.
Rebuild the Series2 span as needed.
Ensure the license URL in the settings page is http://license.voiceware.com/licenses/
Onsite work
Connect a computer to the hotels network and log into the existing system via the web GUI
Navigate to the backups page and take a manual backup
Download the backup to your laptop (if you already restored the backup to the temp system in the office steps 2-3 can be omitted)
Place the temporary server in a secure location that is easy to access. Note that this can simply be on a table. The temporary system does not need to be fully rack mounted.
Power the system on
Attach a screen and keyboard to the temporary system.
Login and issue the command “ip addr” to find the systems IP address
8. Log into the temporary systems web GUI at the IP found in step 7 above
9. Restore the backup from the existing system to the temporary system (if a backup was already restored at the office and a new restore is not needed this step can be omitted)
a. Note: this process might take some time depending on the size of the backup
10. Click “Restart Now” on the yellow banner that appears after the backup is done restoring
Rebuild spans as needed as they are not restored in the backup process
Navigate to Devices --> Hardware Spans
Click Add Span
Check Dynamic
Select the adaptor “enp1s0” from the drop down (note, your adaptor name could differ)
Enter “003018a4c908” as the remote MAC address
Set the Timing to “1”
Set Signaling method to “PhoneSuite Cabinet”
Click Save Span
Scroll down and click Save Span again
If adding more than one span repeat the above process incrementing the remote MAC address by one for each additional span (i.e. 09, 0a, etc.).
Click “RESTART NOW” from the Hardware Spans page, this process should take less than 5 minutes.
The order of the next few steps is important, follow exactly
Set the temporary systems IP to exactly match the existing systems IP address
Click save
Disconnect the existing system from the network
Click the link that says “Click here to restart the system…” on the temporary system
Connect any existing Series2 cabinets to the temporary server using new patch cables
After the system reboots check general phone system functionality as the hotel should now be running on the new hardware.
Note as an alternative a separate network using a temp router can also be created and the system set with the same IP as the existing system. Doing this requires the use of an onsite router and moving the network the laptop is connected to but makes the swap over as simple as moving the network cable from the existing system to the new system without needing to set IPs and rebooting the system.
The next set of steps installs Voiceware 4.2 on the existing hardware while the temporary system keeps the hotel running. This process should take 15-30 minutes depending on network connection speed.
Note also that the system must have an internet connection to complete the installation. Plug the system back into the network only after the reimage process starts.
Place the bootable USB key into a USB port of on the existing hardware (any USB port is fine)
Attach a screen and USB keyboard
Reboot the system
Note a reboot can be done using the very small black power button near the top card tab, or by unplugging the cabinet, or even by using the screen and keyboard to log into the command line and issuing the command “sudo reboot”
During the boot process a BIOS screen will appear, there is often (but not always) a prompt allowing a key (normally but not always F10) to be pressed to enter the boot menu. Press this key and select the USB drive to boot from.
Note in some cases the system will automatically boot from the USB key and in others the BIOS must be entered (by pressing F2 during boot) to select the USB boot drive manually
Reattach the network cable only after the USB boot process is underway.
At this point the system should begin booting from the USB key. Follow the 4.2 installation steps outlined in the section Voiceware 4.2 installation below but stop at step 8 and return to this section.
Log into the existing hardware using the IP found in step 8 of the 4.2 installation steps above
Navigate to the backups page and upload the backup taken from the existing system previously (in step 6 above)
Restore the backup and click “Restart Now” on the resulting yellow banner
Ensure the license URL in the settings page is http://license.voiceware.com/licenses/
Rebuild spans as needed as they are not restored in the backup process
Navigate to Devices --> Hardware Spans
Click Add Span
Check Dynamic
Select the adaptor “enp1s0” from the drop down (note, your adaptor name could differ)
Enter “003018a4c908” as the remote MAC address
Set the Timing to “1”
Set Signaling method to “PhoneSuite Cabinet”
Click Save Span
Scroll down and click Save Span again
Note: if adding more than one span repeat the above process incrementing the remote MAC address by one for each additional span (i.e. 09, 0a, etc.)
Click “RESTART NOW” from the Hardware Spans page, this process should take less than 5 minutes.
The order of the next few steps is important, follow exactly
Set the existing systems IP to exactly match the temporary systems IP address
Click save
Disconnect the temporary system from the network
Click the link that says “Click here to restart the system…”
Reconnect all Series2 cabinets and network cables back to the original system
After the reboot conduct a full system test
Remove the temporary server and all extra cables
Replace the Series2 cabinet door, pickup tools, and secure the phone room
Office Prep Work
Verify license
Upload backup in office to verify how long the restore will take
Rebuild the span
Ensure the license URL in the settings page is http://license.voiceware.com/licenses/
Onsite
Take backup
Connect temp system to network
Power on and find IP
Load and restore recent backup to temp system
Rebuild spans
Set temp system IP to match existing system IP
Disconnect existing system from network
Apply IP address changes to temp system
Note the above three steps could differ if you choose to use a second network and set the temp systems IP to match the existing systems IP before arriving onsite
Ensure the license URL in the settings page is http://license.voiceware.com/licenses/
Test hotel functionality on temp system
Reimage existing hardware
Restore system backup to existing hardware
Rebuild spans
Set existing system IP to match temp systems IP
Disconnect temp system
Apply IP address changes to existing hardware
Conduct full system test
Clean up phone room, remove equipment
Notify hotel staff that upgrade is complete
This method describes upgrading an existing Series2 or 1U rack mount Voiceware system by swapping out the systems hard drive to one with 4.2 preloaded. This method requires the physical removal of the hard drive in a static safe environment. The utmost care should be taken if using this method because there is no temporary hardware available should the equipment be damaged during the swap.
Prepare: for this method you will need
A hard drive that has had Voiceware 4.2 installed on it
A replacement hard drive can be purchased from Phonesuite
Phillips screwdriver
A static free work environment
We recommend a grounded static mat and wrist strap
Screen and USB Keyboard
Check the existing system for duplicate outbound routes, outbound rates, and extensions. Older versions of Voiceware allowed duplicates however a backup containing duplicates will not load into a 4.2 system.
Hard Drive Replacement on the Series2
Log into the existing system and navigate to the backups page, create a new manual backup and download it to your laptop
Power down the Voiceware server using the small black button near the top card tab
If no black button can be found (it does not exist on all hardware versions) use a screen and keyboard to log into the command line and issue the command “sudo poweroff”
Wait a full three minutes
Remove the SBC card
The hard drive is located under the mini ITX mother board as seen in the picture below.
Remove both the power and data cables from the drive
The power cable simple pulls out, while the data cable has a small tab that must be pressed down before the cable can be pulled out.
Turn the board over and remove the four screws holding the hard drive to the SBC board
Carefully remove the now free hard drive
Move the posts from the old drive to the new drive
Carefully hold the new drive in place with one hand while putting in the screws with the other
Reattach the data and power cables
Reinstall the SBC card in the Series2
Reconnect all cables, power cord last
Attach a screen and keyboard
Login and issue the command “ip addr” to find the systems IP address (contact support for passwords as needed)
Navigate to this IP address and log into the system
Navigate to the backups page and upload the backup taken in step 1 above
Restore the backup and click “Restart Now” on the resulting yellow banner
Ensure the license URL in the settings page is http://license.voiceware.com/licenses/
Rebuild spans as needed as they are not restored in the backup process
Navigate to Devices --> Hardware Spans
Click Add Span
Check Dynamic
Select the adaptor “enp1s0” from the drop down (note, your adaptor name could differ)
Enter “003018a4c908” as the remote MAC address
Set the Timing to “1”
Set Signaling method to “PhoneSuite Cabinet”
Click Save Span
Scroll down and click Save Span again
Note: if adding more than one span repeat the above process incrementing the remote MAC address by one for each additional span (i.e. 09, 0a, etc.
Click “RESTART NOW” from the Hardware Spans page, this process should take less than 5 minutes.
After the reboot conduct a full system test
Replace the Series2 cabinet door, pickup tools, and secure the phone room
If the power button connector becomes removed during the swap process below is a picture of where the cable connects to.
Hard Drive Replacement on 1U Rack Mount System
Log into the existing system and navigate to the backups page, create a new manual backup and download it to your laptop
Power down the system using the power button on the front of the system
Note: press the power button only once and wait 30 seconds for the system to fully power down and the green lights to turn off
Label and remove all cables from the system
Remove the system from the rack
Open the case
Note: depending on hardware the screws for the lid are either on the back of the system or on the sides
Locate and unplug the data and power cables from the hard drive
Remove the hard drive from the system
Move the posts (if present) from the existing hard drive to the new drive
Replace the drive and reattach the power and data cables
Replace the lid and screws
Place the system back into the rack and connect all cables, leaving the power cable for last
Connect a screen and keyboard to the system
Power on the system if it did not power itself on when power was attached
Login and issue the command “ip addr” to find the systems IP address (contact support as needed for the password)
Navigate to this IP address and log into the system
Navigate to the backups page and upload the backup taken in step 1 above
Restore the backup and click “Restart Now” on the resulting yellow banner
Ensure the license URL in the settings page is http://license.voiceware.com/licenses/
Rebuild spans as needed as they are not restored in the backup process
Navigate to Devices --> Hardware Spans
Click Add Span
Check Dynamic
Select the adaptor “enp1s0” from the drop down (note, your adaptor name could differ)
Enter “003018a4c908” as the remote MAC address
Set the Timing to “1”
Set Signaling method to “PhoneSuite Cabinet”
Click Save Span
Scroll down and click Save Span again
If adding more than one span repeat the above process incrementing the remote MAC address by one for each additional span (i.e. 09, 0a, etc.
Click “RESTART NOW” from the Hardware Spans page, should take less than 5 minutes.
After the reboot conduct a full system test
Pickup tools, and secure the phone room
Checklist
Check existing system to ensure not duplicate outbound routes are found
Create system backup, download to local storage
Power off the equipment, WAIT 3 minutes before removing cards
Disconnect cables from hard drive
Remove existing hard drive
Attach new hard drive
Reattach power and data cables
Reinstall SBC card
Attach screen and keyboard
Power up and find system IP
Log into the GUI and restore the backup
Ensure the license URL in the settings page is http://license.voiceware.com/licenses/
Rebuild the spans
Full system test
Remove extra equipment and tools
Notify hotel staff that upgrade is complete
If prompted select “Install Voiceware” and press enter. Note this prompt will appear on some installation types and not on others.
At the prompt asking about warning over any data already on the disk select “Yes”.
The system will load the OS, a process that takes about 5 minutes.
Remove the bootable USB key while system is rebooting.
When finished the system will reboot and load a new prompt. The default choice “Voiceware [OEM]” is fine, after a moment the system will boot and present a login screen. Note, it will say Voiceware 4.1 Install, this is okay.
Login using the username “phonesuite” and password “a***********10”.
Issue the commands “sudo -i” and then enter the password again. Next issue the command “voiceware install”.
The system will then install Voiceware, a process taking about 35 minutes.
License & Software Assurance: The system must have a valid license and software assurance active to install. If no license is found (as will be the case with new hardware) the installation process will halt and present the system ID. Contact Phonesuite with the system ID to receive a license and then restart the upgrade process by issuing the command “voiceware install”.
Note that a new percentage indicator has been added to the last section of the download. Also note that at this point if the download is interrupted (system is powered off, loss of internet connection) the system will retain files already downloaded and not download them again. However, the progress percentage always starts at 0%, even on a resumed installation.
When the install process finished the system will reboot. Please note that after the initial installation the system may take up to 3 minutes to fully boot up.
Log into the CLI again and type “ip addr” to receive the IP address, this can then be used to login to the browser or configuration from a web browser.
You will be required to change the CLI password at this point (not pictured).
During the initial boot the screen may display text or messages, this is normal. Simply press enter to display the command prompt again.
Log into the configurator using the username “admin” and the password “phonesuite”.
Navigate to the backups page. From here either upload the Hosted Default Backup downloaded in step 10 above or select the pre-loaded system default backup if the system has two NIC ports available.
Click Restore and then Yes. When the process finishes click Restart Now in the yellow banner and yes again.
Once the system reboots login to the configurator using the default login “installer” and password “Protect6607”
Download and install VMware workstation 16 from Fusion and Workstation | VMware .
Download the Voiceware 4.1 ISO from https://registry.voiceware.com/voiceware/4.1/Voiceware.x86_64-current.install.iso
Select “Create a New Virtual Machine”
Select Install disk image file (iso) and select the downloaded ISO from the local file structure and click Next.
Select Centos 8 as the OS and click Next.
Name the system and click Next.
Set the disk size to 120 GB and click Next
At the Summary screen click Finish.
Select the newly created VMware machine and click Edit virtual machine settings from the right-hand menu.
Select Network Adapter from the left-hand side pick list, then select Bridged network connection. Then click OK.
Select the newly created VMware machine and click “Play virtual machine” from the right-hand menu.
Finished with the VMWare installation. Begin installing Voiceware with step 1 in the Voiceware 4.2 installation process above.
This section covers basic system testing post upgrade along with fixes to common issues along with fixes to issues that might be encountered during the upgrade or installation process.
Testing
Inbound calls to the main number
Check “Block Emergency Use” for the main number and users DIDs (dial plan)
Outbound calls from an admin phone
If calling does not work check:
SIP Trunks are registered
Ensure NAT is checked on SIP trunks and remote SIP phones
System external and local IP subnet set in the settings page items 21 and 22
Ensure system is connected to the network correctly
Check the IP address settings especially the gateway and Name Server IPs (can’t be blank)
Calls to and from a guest room on a Series2
If this fails check:
Cable between system and the Series2 is connected
Span has been built
Span shows a good connection and status
Series2 enabled / disabled check box in the settings page, item 107
Calls to and from SIP based guest rooms
If this fails check:
Device page, phone registration
Physical connection to the phone
If auto configuration was used regen config files and reboot phone
MWI (Series2)
Should MWI not work check the Settings page items 108, 109, and 110. Both 108 and 110 should normally be set to 3 and 109 should be set with seven 5’s (5555555).
PMS connection
If this fails check:
Set the ttys interfaces to 1 and 2 (from 0 and 1)
Swap serial ports then test again
Use the Phonesuite Serial Simulator and a USB to serial device to verify the cables and connectors work
Browser Console access
Note that the front desk might encounter a self-signed certificate warning after the upgrade, they will need to tell their browser that this is okay and to load the site anyway. Also check to ensure the system is on the network and has the connect IP address.
Note also in 4.2 there is a phone in the Browser Console. This can be enabled / disabled in the Users page for each user.
IVR
Emergency calling
Check system date and time
If incorrect set the time zone in the settings page, click save, and reboot the system.
Troubleshooting
Issue: An older systems backup will not restore to a 4.2 system
Solution: Make sure there are no duplicate outbound routes, outbound rates, or extensions. If found delete the duplicates and take a new backup, it should then restore.
On the 4.1 system, you can (as root) "docker logs postgresql
" and find where there are errors.
It's gonna be the first error AFTER "CREATE DATABASE"
Look at the error and then report to Engineering if you don't know what the root cause is.
Issue: Message Waiting Indicators are not working on analog guest room phones
Solution: Check the Settings page items 108, 109, and 110. Both 108 and 110 should normally be set to 3 and 109 should be set with seven 5’s (5555555). If using 4-digit extensions in B12 then set 108 and 110 to 4.
Issue: Browser Console is displaying a trunk alarm for the Series2 span but the span is up and calls to and from a room work.
Issue: The Series2 setting GUI loads but has a warning about being unable to communicate with the Series2 but analog guest room phones are working
Solution: Uncheck the setting 107 Series2 Support
Issue: During the Voiceware installation process the process fails with two errors.
Solution: This is likely caused by a loaded span not matching up with existing hardware. Look at the errors and if they say DAHDI then issue the command “: > /etc/dahdi/system.conf” then issue the upgrade or install command again. Otherwise contact Phonesuite for support.
Issue: System time is wrong, even after a reboot.
Solution: Go to the settings page, ensure the correct time zone is set, and click save (even if no changes are made). Then reboot the system again. If still wrong make sure the system has internet access and check the BIOS clock as well (enter the BIOS upon reboot and look at the time listed).
Issue: SIP trunks or remote SIP phones are not working.
Solution: Ensure that NAT is checked for all SIP devices that need it. Also ensure that the public IP and local subnet are set in the settings page, items 21 & 22.
This is a list of things to check after restoring a backup especially from an older version 1 system.
Make sure the AGI commands for the Wakeup call IVR (SAY TIME ${EPOCH} '') and say extension IVR (SAY DIGITS ${CALLERID(num)} '') were carried over.
Make sure the music on hold works. If not download the sound file, change its name, and reupload it. The default music can also be uploaded if needed.
Make sure B93 is set to 251
Make sure B11 is set correctly (check rooms page)
Make sure these settings match what has been added to the Settings page near the bottom
Note that the serial ports might change numbering from 0 and 1 to 1 and 2. Update the PMS interface serial ports as needed.
Setup / install the SSL certificate as needed (optional)
Using an export tool data can be pulled from a Voiceware system in the format required to import directly into ConnectWare. This is available for Users and Phone Hardware (the only two imports available in ConnectWare). In addition other exports are created with general info on DIDs, call groups, queues, IVRs and others. While these export CSV files can’t be imported to ConnectWare they are useful as guides to rebuild the system.
Note that just because a phone works in Voiceware does not automatically mean its supported in ConnectWare. Recall that all phones in ConnectWare are auto provisioned and you should always check that ConnectWare supports the phones before starting this process.
Method 1
On Voiceware 4.3.0.5 systems:
Log into the Command Line
Issue the command
voiceware export
Open a new tab and enter the URL https://<ip>/exports/<file> (i.e. https://1.2.3.4/exports/users.csv)
Download each file using this method. A list of available files will be displayed when issueing the command but is also listed below.
Method 2
On any system prior to 4.3.0.5, or any version 3 system follow the below process.
Log into the Command Line
Issue the commands
sudo -i
curl -kO https://get.voiceware.com/voiceware/utilities/vwexport
chmod +x vwexport
./vwexport
Open a new tab and enter the URL https://<ip>/exports/<file> (i.e. https://1.2.3.4/exports/users.csv)
Download each file using this method. A list of available files will be displayed when issueing the command but is also listed below.
If the above scrip runs but the files are not downloadable using the above URL then do the following.
cd /var/www/phonesuite/exports/
cp *.csv /var/lib/tftpboot/
You should then be able to download the files using the URL https://1.2.3.4/p/users.csv
Avaiable Files
users.csv (can be direcly imported)
phones.csv (can be direcly imported)
dids.csv
rooms.csv
common_area.csv
ivrs.csv
groups.csv
conferences.csv
Upgrade Series2 to Standalone Server
In some cases upgrading a Serie2 requires the use of an external server in a 1U rack mount case. Below are the changes and considerations that need to be made.
When using an external server to upgrade and use for a Series2 these are the considreations.
Pre-upgrade
Install / upgrade Voiceware on the 1U server before arriving onsite.
Test restore a backup from the Serie2 to ensure it will restore, even if you plan on taking a backup day of install.
Note: if the backup does not restore look for duplicate outbound routes and delete them.
Day of Upgrade
Install the 1U server as near to the Series2 as possible.
A new cable will need to connect the 1U server to the Series2.
Serial connections will also need to be moved to the new 1U server.
Check for external MOH that will also need to be moved to the new server.
Restore a backup to the 1U server before moving any cables from the Series2
Disconnect the Series2 from the network before changing the IP on the 1U server.
Post Upgrade
The daughter card on the Series2 processor card will remain and its not recomended that it be removed.
It can be powered off but it will power itself back on after a power outage. This is okay.
Voiceware Versions
Version Number | Release Data | Notes | Support Ends |
20120918 | 1.0 | Version number is yyyymmdd | Jan 1, 2021 |
10121023 | 1.0 |
| Jan 1, 2021 |
20130222 | 1.0 |
| Jan 1, 2021 |
20130913 | 1.0 |
| Jan 1, 2021 |
20131014 | 1.0 |
| Jan 1, 2021 |
20140314 | 1.0 |
| Jan 1, 2021 |
20141003 | 1.0 |
| Jan 1, 2021 |
20150330 | 1.0 |
| Jan 1, 2021 |
20160215 | 2.0 | Hardware swap required, new OS and Asterisk version | Jan 1, 2021 |
20160503 | 2.1 |
| Jan 1, 2021 |
20170707 | 2.2 |
| Jan 1, 2021 |
20180401 | 2.2.1 | Bug fix minor version | Jan 1, 2021 |
20180612 | 3.0 | Hardware swap required, new OS and Asterisk version |
|
20190328 | 3.1 |
|
|
20190422 | 3.1.417 | Bug fix minor version |
|
20190627 | 3.1.2 | Bug fix minor version |
|
20200601 | 4.1 | Major OS and Asterisk version change, hardware swap required |
|
20200605 | 4.1 | Bug fix minor version |
|
20200706 | 4.1 | Bug fix minor version |
|
20200709 | 4.1 | Bug fix minor version |
|
20200724 | 4.1 | Bug fix minor version |
|
20200731 | 4.1 | Bug fix minor version |
|
20200831 | 4.1.1 | Bug fix version |
|
20200903 | 4.1.2 | Bug fix minor version |
|
20200908 | 4.1.3 | Bug fix minor version |
|
20201005 | 4.1.4 | Bug fix minor version |
|
20201009 | 4.1.5 | Bug fix minor version |
|
20201027 | 4.1.6 | Bug fix minor version |
|
20201204 | 4.1.7 | Bug fix minor version |
|
20210222 | 4.1.8 | Bug fix minor version |
|
20210510 | 4.1.9 | Bug fix minor version |
|
20210901 | 4.1.10 | Bug fix minor version |
|
20220106 | 4.2.1 | New Features and Bug Fixes |
|
???? | 4.3 | Large New Features version |
|
Software Version 20150330
Bug Fixes
Fixed a minor graphical error with the search bar in the browser console
Corrected an issue that was causing upgrades from older versions to fail
Transfer button will no longer be grayed out on 2nd or 3rd pages of rooms on the Browser Console
Stop Bits in the PMS interface section can now be set to 0 and will stay there. Previously the setting would revert on its own.
Known Issues
Special characters cannot be used in the names of IVR states
New Features
Default system backups should now be downloaded upon new system installation.
Custom text can now be entered into the end of the call accounting ICD file from the configurator. Note, please only use this feature under direction of PhoneSuite.
Software Version 20141003
Bug Fixes
A version control document was added for the psip-pms program. It is downloaded automatically when the ICD files are updated from the about menu and can be found in /urs/local/etc/psip-pms.
Emergency call notifications will no longer leave a voicemail, they will simply continue to ring
The park and fast transfer keys now work properly
Corrected a graphics error that sometimes appeared on the top line of the Rooms page
Disallowed the creation of two extensions for a single guest room. This was causing errors with the call billing system
If the first attempt to log into the Browser Console failed the 2nd attempt will now not automatically fail as well
Added a sanitize check to all inputs made when adding new Outbound Routes to the system
Voicemail duration of messages more than a minute long now round properly
Wakeup history in the Browser Console now shows the proper time should the Browser Console user be in a different time zone than the system
Non-real extensions are now not allowed in the Outbound Routs fail option
Clicking edit rules for a time period that does not have any rules resulted in a blank screen, it is now disallowed
Call flags for Outbound Rates can now be removed
Updated the text of the “Value of B25 & B44…” in the setting page to properly reflect the true defaults
New Features
Added external music on hold
Changed the text of some buttons so that the add and/or save options are all labeled the same thing across the system
Improve the look of the Phones page, it now looks like all the other pages (Users, Rooms, etc…)
The system will now restart the Browser Console web service should it stop for any reason
ACL’s have now been turned on by default
Allow the removal of a user’s name on outbound calls (see Outbound Routes in the Dial Plan)
Added externip and localnet to the settings page
New installs will now automatically download and install the most up to date version of psip-pms and psip-stat.
Reorganized the Settings page so things are easier to find
Known Issues
Delete all selected voicemails from the users page does not function properly
Software Version 20140314
Bug Fixes
Moving a guest from one room to another now also brings along their wakeup calls
Removed a broken “export” button in the CDR’s page
Moving a guest no longer leaves behind a “ghost” wakeup call record in the browser console
Removed useless “Default calling restrictions when adding a guest” dropdown in the setting page
Improved process of setting a wakeup call from an admin phone. Process now requires less input and is much quicker.
Changed the email subject from and name on emailed voicemails to “Voiceware”
Copping a state and giving it the same name in two different IVR’s no longer produces an error
Restrict XMLRPC access to local network by default
Moved default outbound routes to the setting page and removed the server page form Devices
When adding bulk rooms and the start and end room number match the error message no longer has visible HTML code
Corrected an issue where moving a user back and forward between departments might cause a loss of Voicemail access. Note that in most cases Departments should not be used and this function will be removed from Voiceware in a later release.
Improved wakeup call reporting in the browser console. In some rare cases the console reports did not show when a guest missed their wakeup calls. This effected reporting only, not the actual wakeup call.
Improvements to the TDMoE spans when connecting to more than one Series 2 box
Entering 0 as the number of digits to strip when adding the default outbound routes now works properly. Previously entering 0 was ignored and it stripped one digit always.
Corrected a broken default outbound route, changed "91NXXNXXXXX" to "91NXXNXXXXXX" (added one X).
Emergency call notifications will not leave a voicemail if they hit someone’s voicemail box.
New Features
Two or more SIP phones can now be assigned to the same room.
Voiceware now downloads all Psip-PMS configuration files upon installation or upgrade
An automatic call can now be generated when a guest misses their wakeup calls (accessed from the setting page)
Added new outbound routes and rates to improve call billing granularity
Added fast transfer and park keys for Polycom phones
Added help pages to all main Voiceware screens and the Psip-PMS configuration screen
ACL’s now turned on by default.
Known Issues
Backups from older versions cannot be restored to systems running 20140314. All upgrades must be done using the built in upgrade feature.
Fast transfer key for Polycom phones does not work properly. Fixed in next version (see above)
Outbound calls from an admin phone always display the users name to the called party Fixed in next version (see above)
Room’s page sometime has a graphical error in the rooms list heeding. Fixed in next version (see above)
Two different extensions can be created for one room, this causes CDR records errors Fixed in next version (see above)
Browser console disconnects at random. Note this should not affect live sites as a patch is being applied as needed. Fixed in next version (see above)
Cannot log into the browser console after a first failed attempt, third attempt should work fine. Fixed in next version (see above)
Copying outbound routes from an excel or word document can add invisible extra characters that will cause those routes not to work Fixed in next version (see above)
Some brands of gateways (some Grandstream) will not work with Voiceware due to a flag on some calls that enables the distinctive ringing feature. If encountered in the field distinctive ringing can be disabled on the server.
Toll free calling is blocked if long distance is blocked.
Setting the stop bits to 0 in the psip-pms screen will not stick on a save/restart. The setting is written properly to the configuration files its purely a comedic issue. Fixed in version 20150330
Software Version 20131014
New Features
Using the “Voiceware-Upgrade” command can now (and should always be) used in the screens program.
Install MFC/R2 E1 codec and rework interface to enhance support for E1 lines in Mexico.
Software Version 20130913
New Features & Bug Fixes
A new tab has been added to the Browser Console to display missed wakeup calls. When a call is missed this tab will begin blinking.
Clicking About or Network Config after the idle timeout has expired will now load the login page correctly.
Users given the Normal (not admin) permission set will now be able to access call recordings in the user’s page.
Improved the search feature for Rooms and Dial Plan to eliminate extra non-matching items to appear.
Setting Weekend and/or Weekday only wakeup calls from the phone now works correctly. This functionality was also improved when setting wakeup calls from the Browser Console. Lastly if a wakeup call can occur the same day it is set up it will do so regardless of the reoccurrence type.
The Drop Admin Privileges link now functions from the Users page.
Behavior of the PMS setting screen has been improved and changing settings should no longer change the port number.
Added the system clock display to the top of the Management Shell screen.
Included version 3.3.3 of the Polycom firmware in all releases of Voiceware. Any Polycom phone configured to Voiceware should automatically download this firmware.
Added additional Polycom phone settings to the Phone page
Prevented the same user from being assigned to the same Phone more than one time.
Added a licenses exceeded warning to the login screen of the Browser Console
Wakeup call history will no longer stick between viewing different guests.
The default block Room-to-Room calling zone has been expanded from 100-399 to 100-799.
Improved the cloning a device function, when cloning the rooms dropdown not populates correctly.
If a filter is set in the Dial Plan screen, then the user navigates away then back to the screen the filter will display properly (previously the filter would remain but the dropdown would get reset).
Outbound routes now sort properly.
Top line of archived guests now displays correctly.
Using the transfer button in the Browser Console no longer changes the buttons appearance.
Restoring a backup now restores Psip-PMS settings correctly.
Appended CID information (using the DID Dial Plan entry) remains on a call transferred though an IVR.
Changing the IP address of a Voiceware system is now properly written to the Polycom config files.
Added a system written language setting in the Settings page.
Corrected an issue where a “Send to Extension” IVR action would not always play ringback to the caller.
The system will now ask for a mail box to be entered if calling the voicemail main Dial Plan type if “enter mailbox number” is checked.
Fixed an issue that would cause archived guests to appear in the DID dial plan entry.
Adding a one off room in the Rooms page now assumes you are adding a guest room.
Folder selection for phrases now works correctly.
CDR table herders will no longer overlap.
Added distinctive ringing to the system.
Added a ten second timeout for the Change Room Status Dial Plan entry so that if a selection is not made within ten seconds the call will end. Also fixed an issue where “-1” was being sent at the room status when the call was ended without a selection being made.
Software Version 20130222
New Features
Guests are now able to clear their PIN numbers.
The Direct to Voicemail Dial Plan type now functions correctly.
Archived guest’s voicemail retrieval functionality improved.
Added the ability to delete phrases in the Sounds page.
Added a red flag to indicate when the number of Devices, Rooms, or Spans meets or exceeds the licenses limit.
Added a new check box to the Settings page to allow or disallow voicemail boxes for guests by default, this can still be adjusted on a per-guest level.
Removed the “Default check-out status” option from the “Add Room” dropdown because this can be set as “NO ACTION” in the settings page.
Changed several system defaults including Room Status Codes, Calling Permission ID numbers, and several options on the Settings page to more closely match what settings are normally needed.
All users (normal and admin) now have access to the emergency call audits in the CDR screen.
When adding a user or room the system will now display an error (and suggestions) when attempting to assign an extension that is already in use.
Added Span Grouping.
Wakeup calls can now be set from Admin phones, so front desk agents can set wakeup calls without using the browser console.
Extensions overlapping with parking extensions can now not be added, the user will receive an error.
Now clicking Settings or Network Config after the system has timed out will now display the login screen (same behavior as clicking Rooms or Devices for example).
Edited the text on the wakeup call delay in the Settings page to help users understand better how to use the setting.
Fixed the broken link “View Users” on the settings page.
Now able to sort wakeup calls by next attempt and last attempt.
Fixed the Local Rate in the Outbound Rates to be sticky and not revert to “Not Rated” every time the entry is edited.
Default Polycom digit map now includes 8000 to allow for voicemail dialing.
The “Call Length” field in the Browser Console now functions.
Outbound Rates now only allow valid characters to be entered.
Fixed an issue where in rare cases the cursor would jump around in the devices name field.
Wakeup calls that occurred in different years will now sort properly.
Answered wakeup calls now show a green checkmark when they have been answered.
Spanish language drop-in voicemail messages now function correctly.
Fixed an issue where clicking a link to show all functional SIP devices did not work when using Firefox.
Added a new dialing permission set to include international dialing.
Fixed a misspelling in the Edit/Create Restricted Zones section.
Night Modes & Drop-in Voicemail messages are now backed up properly.
Fixed the Wakeup Call IVR so that the 10 minute snooze function now works properly.
Other Adjustments
Fail to Ban messages now include the hotel name
Install time of a new image is now about 15-20 minutes longer
Added the ability to upgrade the Voiceware software without reinstalling the entire operating system.
Software Version 10121023
New Features
Added a timed delay between wakeup call attempts field to the Settings page.
Drop-in voicemail messages are now being correctly backed up. Previously drop-in voicemail messages were being ignored by the backup system.
Added a Screen Accept and Screen Reject IVR option. These IVR actions were previously missing making a Screening IVR impossible to set up.
Software Version 20120918
New Features & Bug Fixes
Nightly audit e-mail address and time to run were added under Settings.
LDAP URI and LDAP Login were removed from the Users section.
Removed unused items from the sidebar in the Queues page.
Software Version 20180401 Voiceware 2.2.1
Known Issue
Department names can’t contain spaces or special characters (including “_”).
Custom speed dials cannot be set to 0 or any single digit.
Bug Fixes
Fixed an issue where BFL check boxes would move to other custom speed dials when the entry was saved.
Corrected an issue where the voicemail to email setup page could not be saved after making changes.
Fixed the reporter login (see Queues chapter for more about the Reporter portal).
Software Version 20170707, Voiceware 2.2
New Features
Added a session timeout warning 2 minutes before the session expires.
Added a message to the log in screen if you were logged out due to inactivity.
If a session in the configurator expires it will reload the login screen.
Disabled IAX2 and/or SIP ports when no IAX2 or SIP devices have been added to the system. Ports will automatically open if a SIP or IAX2 device is added.
Many small security enhancements.
Added “Voiceware 2.x” to the about page. (PS-918)
Added Spanish as a language the Browser Console can be written in.
The Browser Console will now always use the extension number for the button in the Browser Console and will display the room name below the guest name. (PS-926)
Added enhanced 911 caller ID and CNAM info for rooms and devices (see rooms or devices chapter for more information) (PS-949)
Added a system notes field to the Settings page (PS-951)
The Console link has been removed as support for the Java applet is near an end. In its place SSH access is available on the HTTPS port (normally 443) (PS-946)
Added a trunk alarm to the Browser Console, clicking on this alarm will display the trunks in alarm and any notes added by the reseller.
Browser Console now displays extension, room name, and guest name at all times.
Voicemail after email now only appears on the Users setup screen after entering a valid email address. (PS-983)
Known Issues
The flashing warning bar at the top of the browser console alerting that an emergency call has been placed will not go away if the emergency call ends. To clear the warning log out and back into the browser console. (PS-917)
Refreshing the phones page after adding a phone entry may result in multiple entries. Simply don’t refresh the page with the red text at the top saying that the phone has been added. (PS-988)
If an admin phone requires a PIN for outgoing calls then the call will be billed as soon as the PIN is entered regardless if the call is actually answered by the called party. This is only of concern in situations where the PMS system is sent costed call data for admin phones (something that is rarely done). (PS-962)
VSP 505 Bluetooth headsets have exhibited behavior that makes it hard or impossible to leave a voicemail when using the headset.
Bug Fixes
Removed German from language list in sounds page. (PS-924)
NAT was stuck on “Yes” in the configurator but set to “No” in the backend database. (PS-923)
Conference recordings now appear correctly. (PS-921)
Corrected an issue where deleting a proxy device might leave records in the database. (PS-919)
Fixed a screen resizing issue for the Psip-PMS configuration window. (PS-927)
Corrected an issue where the CSV import of outbound routes was not working. (PS-936)
Fixed an issue where in a user with admin rights could not turn on or off a user’s “delete voicemail after email”.
Fixed several permission issues that might cause guest or admin voicemails to be unavailable.
Fixed a bug where an outbound route would strip digits twice if both the primary and failover router stripped one digit.
Fixed an issue where the system would attempt other outbound routes that had a less specific digit match if the primary routs trunk was unavailable. This might result in outbound calls being costed incorrectly.
Fixed an issue where if a conference room was created with an admin but no user pin a user could not join (system would ask for a PIN and require one be entered). (PS-994)
Fixed an issue were cloning a user and device resulted in the device missing a port number. (PS-993)
Corrected an issue where the expanded variables check box in an IVR AGI command step was uncheckable once checked. (PS-984)
Fixed an issue where the “Announcement” option for the conference bridge does not save upon setup. (PS-981)
Users can now join a conference bridge with a user but no admin pin (by pressing # when prompted). (PS-994)
Fixed an issue where the Browser Console would use the local computers time and not the server time for determining if the time the wakeup call being set had already passed. (PS-1005)
Software Version 20160503, Voiceware 2.1
New Features
Qualify timer for devices is now set to 60 by default. (PS-896)
Added check boxes for the dial plan entries so that they can be deleted in mass (PS-892)
Added a BLF check box to the custom speed dial options in the Polycom phone setup page. This can be used to add BLF functionality for users and parking spots. (PS-886)
Removed the text “User:” from the Polycom phone display to free up screen space (PS-884)
Added a new check box to the About --> Updates tab to allow an installer to update the Psip-PMS program. (PS-901)
Firefox is ending support for the Java application that allows access to the Linux backend via the web GUI. As such we have added the ability to access the SSH backend by pointing Putty to the IP address and HTTPS port (i.e. 443, or other translated port). This allows SSH backend access without needing to open port 22 or add a port translation to the firewall. (PS-898)
New PRI Signaling Methods have been added to allow a PRI connection to another PBX. (PS-909)
Room names rather than extension numbers can now be used as room identifiers in the browser console (setting 28 in the settings page).
Calls recorded using the *1 call record feature will now be emailed to the user like a voicemail. (PS-614)
Caller ID info for active call now displays in the browser console (above the time / date). (PS-593)
Rooms can now be added via a CSV file (see the rooms page for more info).
Browser Console new features include:
Improved experience for mobile users. The buttons and menus will resize if using the browser console on a phone or small tablet. (PS-691)
Wakeup calls can now be setup for guests based on affiliation. (PS-751)
Added quick set wakeup call button for one-off and bulk wakeup calls. (PS-691)
Bulk drop in message to guests by affiliation or to all guests. (PS-746)
Added software version to the browser console screen above the time. (PS-907)
Bug Fixes
Corrected an issue where help files were being downloaded into the incorrect location and thus would not update. (PS-897)
Fixed an issue where “0” was not allowed as the number of digits to trip when using the add defaults function in the Outbound Routes. (PS-899)
Fixed an issue where DIDs starting with 0000 could not be added using the bulk add feature for DID dial plan entries. (PS-895)
Fixed an issue where if a DID was assigned to a room and that room was deleted the DID would get “lost” in the database and become unusable (also PS-895)
Corrected an issue where both the local and server time in the Browser Console were always the local time of the computer accessing the system. (PS-894)
Changed wording from “Empty Room Calling Permissions” to “Empty Room Calling Restrictions” to more accurately reflect what the drop down does. (PS-883)
Can now add more than one custom speed dial for Polycom phones. (PS-882)
The browser console now correctly displays the number of guest voicemails (if any). (PS-880)
Removed the SMS Text option from the IVR action options because the feature did not work and most carriers allow you to email someone at their phone number and have it translate to a text message. (PS-876)
The Queues help page now works correctly. (PS-875).
Reminded system of basic curtesy and if user’s extension is set to “Hangup” upon timeout the system will now say “goodbye” before hanging up. (PS-874).
Someone with user level permissions can now view the Dial Plan and CDR (if allowed in the users page) pages. (PS-873)
Changes to the Device in the PMS interface page will now save correctly. (PS-879)
Corrected an issue that caused the paging call group to not work at all. (PS-912)
Fixed an issue that caused paging to fail for all types of phones.
Known Issues
Removed non-working attended transfer button from the Browser Console. This button is not going to be put back for some time if ever. (PS-850)
Software Version 20160215 Voiceware 2.0
New Features
Basic Shared Line Appearance
Reverse Caller ID
Failover outbound routes
Expanded Polycom Settings
Dialing permissions / restrictions based on rate table
Wakeup call retry after system outage
Pop-out log viewer
Did pool and bulk add of DIDs in the Dial Plan page
Proxy box feature for hosted or physically distant systems
Auto trunk tester
IVR play sound state allows you to listen to the sound
Hot desk support
User based time zone settings
CID routing rules
New IVR options (email and switch)
Software Version 20190627, Voiceware 3.1.2
Bug Fixes
Fixed an issue where a user could not enter a “/” in the device names for local type devices. PS-1173
Adding a local type device (that has no password) will no longer complain about having a week password. PS-1175
Fixed an issue where a user could not navigate away from the rooms page until the TDM port info loaded (a pain when the span was down). PS-1174
Removed the PMS Connector update checkbox from the updates tab in the about window. This was done because updating the PMS connector on a 3.X system will cause Psip to stop functioning. PS-1169
Known Issues
B32 always displays 8 regardless of what its set for. If it’s changed to 8X it will set correctly on the TDM side but not display as such in the GUI.
PRI DIP switch display in the B Feature programming page does not display actual settings of the DIP switch.
Time zones set for a user don’t always propagate to the SIP phone due to the different ways in which SIP phones set time zones and what they accept as valid options.
Pushing TDM programming via the Manage Series2 Files screen might result in a false error saying that the programming was not pushed and to try again later. Always check after the push to see if the TDM programming was updated correctly.
Dialing **8XXX will result in only the last three digits being dialed (i.e. if you dial *8405 only 405 makes it to the dial plan). There is a work around for this issue, contact support for help.
New Features
Added patch version to the version number. Version number is now Major.Minor.Patch (i.e. 3.1.2).
Software Version 20190422, Voiceware 3.1.417
Bug Fixes
Fixed an issue where billable calls were not being passed to the PMS.
Fixed an intermittent issue where the autoconfiguration for Polycom phones would not work, this also resulted in the IP address for the phones on the Phones page to be missing.
Software Version 20190328, Voiceware 3.1
New Features
Expiry time for Device Registrations lowered to two minutes. This should aid in 4G failover or other general IP change situations. PS-1124
Added non-guest rooms to the printable extension list. PS-1098
Added ability to bulk add Proxy rooms. See the rooms page for more info. PS-989
New Vtech zero touch feature available for the newer Snom based models (see the Phones page for more info).
Added the ability to add an override server IP for the phones page via the settings page so the override IP is used by default on all new phone entries. PS-1080
Added the ability to set a trunk prefix (or no prefix) to the add default outbound routes page. PS-1069
A very annoying warning will appear if you create a device with a weak password. PS-1120
Added the Ring in Use check box for Queues (see Queues chapter for more information) PS-1071
Removed departments from the user’s page (they are not currently being used for anything).
Email field (Settings item 23) no longer requires you to also enter the hotel name, it will use the hotel name from item Setting item 2.
Adding a park key on Polycom phones will no longer display a park key at all times, it will only display if a call is parked.
The TDM port number will now be displayed in the rooms page for virtual device rooms.
Known Issues
Drop in messages cannot be overwritten using the Sound Recorded dial plan function. If a change is made to the sound file you must change the sound file the drop-in message points to, save it, then point it back, and save again. PS-1132
Reboot button on the phones page will only reboot the phone if changes have been made to require a reboot. This feature continues to only work on Polycom phones connected to a local (not hosted) system.
If while on the guest details page the paired extension receives a call the transfer button on the details page will not activate. Transfers can be completed from the Rooms page if needed and this issue does not happen if you start on the Rooms page, receive a call, and then go to the Details page. PS-1144
Adding an and sign (&) to a username or call group name breaks Polycom directory files. The only fix currently is to remove the & sign, regen the configuration files, and reboot the phone.
Bug Fixes
*8 can now be added to the custom speed dials for Phones. PS-1123
Placing a guest in DND via the Browser Console will no longer prevent broadcast messages from being received. PS-1061
Forwarding a voicemail will now retain all meta-data on the message. PS-1118
Issues where phones lock up if they are in a complex looping call group setup should be reduced if not eliminated altogether.
Issue with large systems having MWI lamps on analog phones being out-of-sync (or behind) should now be fixed.
Software Version 20180612, Voiceware 3.0
New Features
Added a warning if a user attempts to set the fail extension of a call group to itself. Note this behavior should be avoided. PS-1055
Added the pickup group and SLA a device is assigned to the main devices list. PS-1004
Phones template files are now included in the backup file. PS-1049
The files required for hosted backup are now automatically downloaded and applied. PS-979
Added the ability to have the system push CDR records in near real-time to a hosted repository for use by call reporting software (such as Genesis).
Removed the character limit for the override server IP in the phones page to support a FQDN not just an IP address.
Known Issue
If a guest is in DND (set via the Browser Console) they will not receive broadcast messages. PS-1061
The file audit_log is not being rotated and thus might eventually fill up the disk. This issue is being manually fixed on all shipping systems post 9/1/2018.
Some systems shipped were referring to a local repository to download system files and updates, this would cause errors in the field because the local repository was missing. Systems shipped post 9/1/2018 have this corrected and any live system can correct this issue with the command “sudo rm -f /etc/apt/apt.conf”
Bug Fixes
Fixed an issue were the emergency CID or CNAM from a guest room will be set as “-1”. PS-1060
Corrected an issue where 0 could not be set as a custom speed dial for phones. PS-1058
Users logged into the configurator can now turn off voicemail to email. PS-1054
Corrected an issue that made it hard to change the sound files used for Queues. Before this fix Asterisk might have needed to be restarted to effect changes in sound files for the Queues. PS-1051
Fixed date sort for wakeup call audit logs in the configurator. PS-1042
Fixed an issue where if editing a hunt group a user is selected in the drop down but not added to the group and then the save button is pressed an error screen appears. PS-1024
Corrected minor text formatting issue that appeared in the warning box when resting a user’s voicemail. PS-1008
The settings page now enforces a 15-character limit on the syslog server setting (56). If a user put more than 15 characters in this field it would not then be possible to add a Phones entry. PS-1006
Removed the folder option when copying an IVR. PS-1003
Removed “Polycom” from the settings page as the settings there can now apply to more than just Polycom phones. PS-996
Fixed an issue where a user could dial *72 to setup call forwarding but then not enter a forward number. The call would time out and end and then if anyone called that user they would receive an error message. PS-1002
Conference bridges can now not be setup with only an admin pin. PS-994
Version 20200601, Voiceware 4.1
New Features
Directed call pickup [PS-1159]
Enhanced information for registered SIP devices [PS-1052]
Users can now be granted permissions for each page individually [PS-1074]
Staff can now play a drop in message from the Browser Console before dropping it into a guest’s voicemail box. [PS-1067]
Notes can now be added to the emergency call alert box that appears in the Browser Console [PS-1146]
The system now warns you when you are leaving page with unsaved information [PS-1147]
B93 can now be toggled from the Series2 programming page. [PS-1149]
The system now pulls and stores backup from the TDM side automatically [PS-1150]
SLA devices no longer appear in the device list when creating a call group (because adding them to a call group will not work) [PS-1151]
Connecting a USB cables to a UPS will now display an alert on the Browser Console should the power be lost to the UPS. Includes an estimate of remaining battery. [PS-1153]
Time zones can now be set via the Settings page. [PS-1154]
Devices can now be part of more than one pickup group [PS-1160]
An email account can be added to the system via the GUI (previously a bash script had to be used). [PS-1162]
An outbound call can be generated from the GUI [PS-1168]
The Browser Console has a soft phone built in and no longer requires the user to add their extension upon logging in (it will automatically pair to that users extension, this can be changed as needed). [PS-1186]
New voice prompts can be created in the Sounds page using text to speech. [PS-1191]
Dial Plan has been alphabetized [PS-1192]
Help text on most pages has been improved [PS-1198]
Various text and button name improvements.
DID’s can now be added in bulk even if non-sequential. [PS-1214]
Bug Fixes
Fixed an issue where editing a phone entry would result in the missed calls check box getting unchecked. [PS-1223]
Know Issues
The *1 call recording feature only works from phones that started (not received) the call.
Version 20200605, Voiceware 4.1
Bug Fixes
Fixed an issue where Asterisk would not start due to a permissions issue. Note: no system was shipped with this version.
Fixed an issue with SSL certificates not being able to be installed
Version 20200706, Voiceware 4.1
Bug Fixes
Added an alias for Sertest
Fixed an issue where 3.1 backups were not carrying over the default outbound CID from the settings page
Fixed a JSON error when editing the fail route
Fixed the IVR preview from the dial plan page
Version 20200709, Voiceware 4.1
Bug Fixes
Fixed an issue where the default backup was not pulling from S3 correctly.
Version 20200724, Voiceware 4.1
Bug Fixes
Fixed delete multiple devices error
Fixed a text error in the from line on the show CELs page
Fixed error on show CELs page when there were 0 calls in the CDRs page
Version 20200731, Voiceware 4.1
Bug Fixes
Fixed issue where you could not set a system back to DHCP
Disabled the PMS interface on new installs and fixed an error where the PMS interface could not be altered
Version 20200831, Voiceware 4.1.1
Bug Fixes
Fixed an issue where calls were not being processed or posted to the PMS
Fixed an issue where the option to allow callers to an IVR to call directly to a guest room
Version 20200903, Voiceware 4.1.2
Fixed all issues regarding call accounting posting to PMS
Version 20200908, Voiceware 4.1.3
Fixed a few remaining issues with call accounting posting to PMS
Version 20201005, Voiceware 4.1.4
Fixed an issue where outbound calls on admin devices can in some cases bypass call restrictions
Updated the text for analog rooms when adding rooms in bulk (no bug, just for clarity)
Fixed several spelling errors on the settings page and cookies warning page
Fixed an issue where the clear fail2ban button did not work
Added smatctl, nano, and nmap as default installed packages
Fixed an issue were the custom email account was not being included in system backups
Updated the network config page to look like it did in 3.1
Fixed a page error when loading bulk MAC addresses (feature worked, just a display bug)
Fixed a display issue where the P-Asserted-Identity check box would not stay checked (feature worked, just a display bug)
Fixed an issue where guest calls would show "s" as the destination
Version 20201009, Voiceware 4.1.5
Fixed an issue were missed wakeup call notifications would not ring to the front desk
Version 20201027, Voiceware 4.1.6
Bug Fixes
Fixes email notifications
Fixes 'EMERGENCY' notifications as well (call group, bc)
Known issues
BC notifications do not appear for non-room emergency calls (e.g. front desk or GM dialing 911)
Version 20201204, Voiceware 4.1.7
Bug Fixes
Fixes Series2 programming push mechanism.
Fixes Hosted server Music on Hold class issue. To fix previously broken classes, recreate them or just go into the class and hit "save" without making changes.
Fixes duplexing issue that created timing slips on some Series2 installations
Fixes visual bug showing emergency caller-id for a room as '-1' when it's not set.
Fixes "Fatal error: Uncaught Error: Call to undefined function reboot()" when using "Unconfigure Endpoint" option in Endpoints screen.
Fixes the synchronization of digit map changes from option 55 on the settings page to all configured phones
Fixes forwarding to timeout extension on user extensions, even if user answers
SIPNP (PnP Provisioner) will now function again.
Swap out NTPD for Chronyd to fix time sync issues
Fix Queue BLF Functions
Performance Enhancement
Use Phonesuite/SATS STUN servers for enhanced call setup time
Future upgrades will be able to use the fetch-first method
Known Issues
IVR Set Night Mode does not work, it always reverts to Day Mode. For now set Day/Night mode via Browser Console.
The Group Code GRP does not function and has not likely functioned for some time.
Version 20210222, Voiceware 4.1.8
Bug Fixes
Fixes issue regarding Night Modes not being able to be set via the static option in the IVRs
Fixes issue not allowing less than 3 digits for room numbers in wakeup setup extension types.
Fixes issue where SLA can cause a SIP deadlock when last of multiple parties places SLA caller on hold.
Fixes issue where PRI deadlocks on analog to analog calls, especially when using analog => call group w/local channels pointing to analog rooms. (Upstream fix from ASTERISK-28605 for 16.8.0)
Fixed an issue where special characters get stripped out of a new DID when adding a user.
Fixed an issue where special characters get stripped out of the DID forward state.
Fixed an issue where "!" could not be added to the dial plan. Note that ! is 0 or more digits.
Fixed an issue with Agent Availability reports (in reporter)
Fixed using 'date ranges' in the reporter
Known Issues
Guest calls to admin phones that are forwarded to an offsite number might be billed
Version 20210510, Voiceware 4.1.9
Bug Fixes
Fixed a bug that was causing call recordings to not be purged correctly and thus leading to a full hard disk.
Fixes issue preventing proper greetings management for voicemail boxes from both the Dial Plan Extension edit page, and the voicemail greetings management page. (VW-124,VW-125,VW-137)
Fixed an issue where the show missed calls check box for endpoints did not save the first time (VW-141)
Fixed an odd warning when you attempted to save an IVR switch action with no case value. (VW-118)
Fix an issue that can occur when a failed CSV upload leaves behind broken rooms_devices entries, causing all devices to become ineligible for group memberships.
Fixed an issues with the recorded calls beep not working.
Fixed an issue where *1 would not always record the call (VW-76)
Fixed an issue where adding 0 to a speed dial for endpoints would add any other extension that had a non-numeric character as well. (VW-147)
Fixed an issue that could cause the move_room xml-rpc function to seemingly fail due to a race condition and unspecified room id. (VW-158)
Fixed an issue that would cause an error when trying to unarchive a user (VW-163)
Known Issues
It is possible for a guest to be incorrectly billed for calls that are the result of a forwarded admin phone. (VW-120)
Calls will go to voicemail regardless of timeout action when sent to a user from an IVR state of "send to" when "user" or "extension" is used. (VW-161)
New Features
New System Default Rates and Routes (VW-138)
New Weather IVR sounds added (VW-145)
Removed GSM as a default codec for devices and added G729 and G722. (VW-67)
Restore progress is logged to system logger (view with journalctl -f | grep -i Restore )
New Polycom firmware for newer VVX phones
Version 20210901, Voiceware 4.1.10
Bug Fixes
Fixed an issue where fail-over outbound routes did not work
Version 20210927, Voiceware 4.2
Bug Fixes
Fixed an issue that could prevent endpoint saving on systems with large amounts of extensions. (VW-170)
Fixed an issue that prevented drop-in-messages from working via PMS commands (VW-171)
Fixed an issue with restoring v1.0 backups relating to tftp
Fixed an issue that would cause the checkbox for the default caller-id in the settings page to be ignored if there was a caller-id in the default caller-id box. (VW-173)
Fixed an issue that prevented sounds from being overwritten using the sound recorder (VW-172)
Better log file management for psip-pms (VW-23)
Change display of Polycom version numbers to reflect updated firmware (VW-184)
Fixed an issue where a guest might be billed for a call to an admin phone that was forwarded to an outside line. Note in this case the call is not costed. (VW-120)
Fixed an issue where non-guest rooms appeared in the drop down of rooms when manually adding a guest.
Fixed an issue where the system default trunk could not be selected when adding the default routes.
Removed DAHDI devices from the direct to device Dial Plan entry destination.
Fixed an issue where email from the IVR did not work.
Corrected an issue where uploading a new sound file to the default MOH class would cause the default check box to become unchecked.
Product Enhancements
Added all default codecs to rooms added with CSV instead of just uLaw. (VW-174)
Stopped restoring spans from versions prior to 4.x (VW-180) and display a warning that spans may need to be recreated if backup restored from such a version (VW-72)
Re-ordered sounds in the Drop-In Message selection to be in alphabetical order (VW-74)
Improved operation of call posting to psip-pms and removed failure points
Added the ability to reboot Yealink and Vtech SNOM phones remotely (VW-178)
Added dynamic location routing for emergency call dispatchable location information (NG911)
Added a log rotation for Psip log files. (VW-23)