- Multiple PBXs can be linked to eachother, so the Users of the linked PBXs can querry the Presence state from eachother. Furthermore contacts from linked PBXs can be added to the buddy list etc. Important: In this scenario, you don't use the number of the external contact rather you use the main number. This means, you can only find and subscribe to users which have a main number configuerd.
The configuration/connection has to be done via CLI (DB entry + reboot)
- Teams can now be integrated in a way so the Presence state will sync (just read).
For this there are no Voice licences from Microsoft needed. Additional, the FQDN of the PBX does not have to be changed for the sync and the user names do not have to match the Teams user names. You just have to activate the mode "Teams only" in the PBX settings, add the teams domain, follow step 8 form the our wiki site MS Teams and get the respective UUID from the Azure AD and add it in the PBX under "MS Teams UUID". The point of this kind of connection is to set the user on "busy" while they are in a Teams meeting. (There are many customers, which espacially because of this only want to use "one tool", so they don't get disturbed while they are busy in Teams)
- Bexio link: Adress book sync: Adress book contacts from Bexion can be automatically synchronised with the Adress book of the PBX (One way communication: Contacts from Bexio will be added to the PBX, newly added or changed contacts dont get synchronized from the PBX to Bexio)
- Bexio link: Calls in time recording: Calls, including notes, can be saved in Bexio and
can be billed to the customer. Attetion: for this is a new version of the wwphone
client needed, which isn't released yet.
- You can now define in Mappings that a number and after connecting the call and
- postdialling via DTMF can be done (this only works with calls to external numbers!) The definition takes place via number, DTMF
- The Zabbix agent does not have to be started via start script and configuerd
manually, rather it will be automatically configured and started as long as in the
config_table in the column zabbix_server server IP has been entered
- The QR code now has a grey blackground not a black anymore, which will make
it easier to scan the QR code for some mobile devices
- For conference calls you can now set if a accoustical signal should be played or
not when someone enters or leave the conference call
- 2 new options for Queue, "time between announcments refers to the end of the
announcement" and "announce the place on hold"
- The creation of the name for the buddy list for wwmobile now is done according
the same criteria as wwphone/CTI. If there was now nickname, it was possible that
the user was named differently in the buddy list of the app as in the wwphone/CTI
- CTI/wwphone gets notified when the Adress book changes so that, with a random
timout so not all clients try to load the adress book at the same time) they can
reload the adress book. Attetion: for this is a new version of the wwphone client
needed, which isn't released yet.
- Now, when you change the username of a user, the device linked in Provisiong
gets requested to reload the config BEFORE the asteriks config gets changed.
Through this the Notify for the reync is send with the correct user, which the device still knows. It should not be needed to restart the device anymore that it will get the new config and log in with the correct user name. Attention: If you change the username multiple times in a short ammount of time, it can still be that the resync won't work
- Passwords won't be directly displayed in the web interface, you have to click the password box to see it
- If a subscription to Microsoft teams is not possible, because a UUID isn't valid anymore or wrongly entered, the UUID wil be removed automatically from the request and the request will be tried again
- For RTX and Snom multi-cell DECT systems there are new firmware verisions available. Which version the senders should get can be configured in the Provisioning settings
- wwphone/CTI 3.3.0 (Windows, Linux, Mac OS)
- Asterisk 16.16.1 contains multiple security fixes (none that can be used for more than DOS and by unauthenticated users) and a fix for a memory leak, which could lead to out of memory when the user uses wwphone/CTI and is muting the microphone regularly.
- As an administrator, you can send the employee an activation link from the web interface for wwphone/CTI or wwmobile.
- A push is now also sent in the telephone service status
- The presence status can be saved under wwphone/CTI from version 3.2.7.
- The presence status is retained when the switchboard and/or the presence server is rebooted.
- The new DISPLAY_DIVERSION_ENABLE option is now set for Panasonic telephones.
- It is also possible to add only signalable numbers under the wwphone/CTI subscriptions. This, of course, only happens in the mode "All (incl. outgoing)".
- On hypervisors with many PBXs (400-600) two cron jobs per minute (since KVM301) could lead to the hypervisor allocating too little CPU time to individual instances in the short term and therefore RTP packets piling up. A long-term optimised approach has been integrated in this version.
- Only RTP packets <300 bytes are stored when tracing with RTP (signalling packets still unlimited).
- Whether traces need to be cleaned up is checked every minute, so that even on systems with very high RTP traffic, traces are deleted in time.
- A new maximum connection time for calls to the Microsoft Graph API via curl library is set to avoid the API call from hanging too long in the case of a communication error and subsequent calls being processed with a delay.
- The bandwidth allowed per stream is now switched back up when conference participants with video stream enabled leave the conference (and then dial back in without video, for example). Previously, the quality was continuously reduced as more users with video participated, but not increased again when video users leave.
- Template for snom phones 7xx was adapted.
- wwphone/CTI 3.2.9 (Mac OS).
- wwphone/CTI 3.2.8 (Windows, Linux).
- wwmobile 3.5.9 (iPhone)
- wwmobile 3.5.7 (Android)
- Sometimes the Asterisk hangup script is not executed completely, which could very rarely lead to PUSH clients staying in “ringing” mode. This very rare phenomenon causes Ringing to appear in the status of employees. This will now be fixed after 5 minutes at the latest.
- Device selection in Chrome for web conferencing only worked if the device had a camera connected. In Audio-Only setups, we could not select devices directly in our WebRTC client. This has now been fixed.
- The switchboard can be linked to Teams and thus operated as a hybrid solution. Presence status is matched with Teams (only from Teams to PBX, not vice versa), so employees who don't use Teams can still see if someone is available or not. All system features can be used as normal, Teams users simply use Teams as a client instead of a normal hardphone or softphone.
- Performance optimisation when reading an address book (large amounts of data >5000 contacts) could block the sip.db for several minutes.
- SIP trunk password is no longer displayed in the web interface.
- wwphone/CTI 3.2.5 (Windows, Linux and Mac OS).
- If you picked up a call with the app using the "pick up active call" function, you could not transfer it.
- The subject in infomails about missed calls was not UTF8 encoded. This could mean that the subject of callers with umlauts found in the address book was not displayed correctly in certain e-mail clients.
- If a telephone was no longer assigned a profile in provisioning and a user was deleted who did not appear in any provisioning profile, the corresponding device was deleted.
- The info e-mail that the http/2 client is rebooted to control the Apple Push Service is now sent in a separate process to avoid a hanging Sendmail process from preventing the reboot of the client.
- Optional 2-factor authentication for admin users.
- Group logins can be blocked so that users cannot log in or out themselves.
- Groups can be completely hidden, so they are no longer visible in the clients.
- User logoff can be defined in such a way that at a lower threshold value no further logoffs by the users are possible (automatic logoffs at timeouts in queues or manual ones by the admin are not affected by this).
- Users can be blocked from logging in and out of a group (the status is still displayed in the client).
- Conference number: You can define one as a "dial-in" and then assign each user their own conference room, so each authorised user has their own conference room, which is external with one telephone number for the whole system.
- Adjustment of some provisioning parameters for the Panasonic TGP600 on request of a reseller.
- Chrome on Linux and Windows now supports the selection of audio and video devices for WebRTC conferences directly in the WebRTC frontend.
- Chrome could not correctly join the conference in "Split screen" mode if "Join without audio" was enabled at the same time.
- Special circuits: Negative filters are now available. Numbers or number blocks (with * at the end) can be defined separated with a semicolon, for which the special circuit should explicitly not apply. The filter and the negative filter can be combined, with the negative filter being evaluated first.
- Network call list for Yealink desk phones: Yealink desk phones can be provisioned to retrieve the call log from the PBX instead of keeping it locally. So you have more info available (not only source but also destination number, addition with name if stored in the phone book and tel.search lookup) and also the indication of whether a call was missed or not is more reliable.
- Interface for CTI client to manage conferences and conference invitations.
- The call limit can now be set so that outgoing calls from group members are not included in the count.
- The setting of the call forwarding, special circuit and group registration status now runs directly via presence server and thus much more efficiently.
- WebRTC now runs on port 443 (except for RTP), so no additional port is needed.
- WebRTC note for Safari: Safari users are reminded that there may still be a problem or two with Safari.
- The web interface now displays the corresponding web link for conference invitations.
- WebRTC labels under the dial-in logos.
- Free-seating with Mitel/Aastra desk phones did not work correctly (phone was not automatically reprovisioned).
- Only listening mode under mobile Safari did not work (under mobile Safari the microphone is now tapped in this mode, instead of a "virtual device", but muted by the PBX).
- In the web frontend, calls to numbers that you have answered yourself but whose status you have not "subscribed to" are now displayed in a new way.
- Calls to numbers with on-hold queues which you have answered yourself but whose status you have not "subscribed to" are now displayed in the call history.
- In certain connection scenarios, there was one-way audio after connecting via mobile app as long as the compressed codec was enabled in the app (e.g., if the call was put back on hold). It's fixed now.
- If someone disabled the camera transmission (black image displayed), the video was incorrectly not counted towards the number of active video windows, which now no longer messes up the layout.
- For KVM301v4 and higher, patch for Asterisk, because Asterisk sends internal protocols in the SDP and thus the SIP packets are unnecessarily enlarged. This patch removes the internal packages. Is only relevant for routers with NAT and "don’t fragment" and the use of many protocols.
- The web interface has undergone a facelift. Resellers can set the displayed logo and background images themselves and influence certain basic colours.
- PUSH is also supported in on-hold queues.
- In the CTI/wwphone Subscriptions, there is a new mode called"All (also outgoing)". With this mode, you no longer see all incoming calls and outgoing calls that you have made yourself, but also outgoing calls that have been made by other employees via this number.
- You can choose whether an announcement or mailbox should always come at the end of the dial plan (if the call remained unanswered) or, for example, only if the participants in the last step were not busy (or only if they were busy).
- The label of a number, which is shown on the phone display, can be set directly in the web interface.
- A header can be set in the web interface, which is sent to the phones with a certain dial plan. This can be used, for example, to use different ringtones for different numbers (note: terminal-specific!). This parameter should only be configured by experts, as it is possible that faulty headers can cause the terminals not to ring at all.
- Admin permission can be set to see details of connected OpenVPN clients under Locations and manage certificates.
- The Flex Core menu now shows when a wwflex user last logged into the system.
- For conference numbers, the sender name can be set.
- The telephone number is now displayed in the international format in conference invitations.
- If an INVITE is sent to a user via PUSH and the user has not logged on to a hardphone or softphone at the same time, it is now also possible to see that the extension is ringing and it is now also possible for employees to pick up such a call.
- "Signal busy when ringing" is now also taken into account in the PUSH scenario without parallel logon of a hardphone or softphone.
- Certain versions of the Exchange Server could not handle the sent conference invitations correctly and displayed the duration of the conference incorrectly.
- When editing the invitation text and inserting line breaks, the complete text was not displayed in the invitation.
- If, due to network problems, a CTI client opened several unsuccessful connections and did not terminate them correctly, it could take a few minutes until the connections were terminated on the server side and thus login slots for the user existed again.
- The parameter NETWORK_CONFIGURATION_SERVER for RTX multi-cell DECT systems was not set in the VPN scenario and had to be set manually on the transmitter.
- Panasonic devices in the HDV series could not handle offered SRTP as soon as one wanted to make a conference call or connect the calls. The telephones are now provisioned so that these scenarios work.
- In certain constellations, the download of large files (call recordings, traces, etc.) could lead to the PBX getting into an out-of-memory situation. The download method has now been revised.
- Firmware for RTX is only offered via https, so also with VPN box, to align with the Snom multi-cell DECT system. This makes the two systems more similar in behaviour
- Direct call forwarding now actually behaves exactly like a LINK (not just in certain scenarios).
- For the decision whether a mobile device is logged in or not, i.e. the yellow icon should be shown, only the token for the VoIP push notifications is taken into account for iOS devices and no longer the second token for the notification of missed calls at the same time. Because if someone deleted the app and reinstalled it, but disabled PUSH notifications for missed calls and consequently the system never sent notifications to this token, the system never even became aware that this token was no longer valid and thus a wrong status was sometimes displayed. This is the case if the current token was logged off, but the old one was still logged in!
- The iOS app can now automatically "cancel" signalling for a call (e.g., if already busy or if rejected by the user). This prevents a "ghost call" from being displayed for a short time in certain constellations.
- Migration of the interface to the Apple Push service on the new HTTP/2 interface, the binary interface will be discontinued in November 2020, as Apple announced in passing in November. This means; by November, all customers using the iOS app should be migrated to a KVM300v3 version.
- Linking version 3.0.25 of the clients (on Mac OS and Windows compatibility with Jabra Engage headsets).
- additional icon in the CTI client/softphone showing that no telephones are logged in, but a mobile phone is (yellow icon ). This yellow icon is only visible if the mobile app has not been deactivated with the slider at the top. This makes it easy to determine whether a user could be reached via the app. If a normal telephone (alone or simultaneously with mobile) is logged in, the green handset appears as before.
- In the special circuits, a status can be set for a user when a special circuit is activated in "Always" mode. If it is deactivated, this status is deleted again (unless it has been modified/overwritten in the meantime). In this way, specific status messages can be stored for individual special circuits.
- New function "Send extended info to telephone about calls to other users" in the user administration (applies per user): this function ensures that devices that support this and have been configured with the necessary options display not only that someone is ringing, but also who specifically is calling the employee. (Attention: after activating or deactivating this option, it is necessary to log out of the subscription and log in again, which can be achieved by rebooting the telephone, for example. Until then, the previous value applies! (Mere automatic renewal of the subscription is not sufficient).
- Newly appeared special circuits for which a user has been given the access rights to activate/deactivate appear within a few seconds in the CTI client (requires version 3.0.24) and also disappear again immediately as soon as the permissions are withdrawn. Previously, the adjustment (visually, but not rights-wise) applied only after the next login (or after modifications, which caused a complete re-importing of the user list).
- The set text status can now also be queried through the web API accessed by the mobile apps. Thus, in upcoming versions of the apps, the out-of-office messages can also be displayed as in the desktop client.
- WebRTC video conferencing via browser.
Conferences can be scheduled, assigned a temporary pin code, and invitations sent by e-mail. Participants can be muted in the web-Gui. The conference frontend in the web-Gui refreshes every few seconds.
- Special circuits can now also be applied to internal users (exclusively or combined with external numbers).
- Filter under telephone numbers, with which you can display the unused/free numbers.
- When creating a user via the quick wizard, the entry for PUSH: username is now also automatically generated for direct numbers.
- When renaming users, not only the entries for the user in the dial plan are adjusted to the new name, but also for PUSH: Username.
- Provisioning of the Panasonic TGP600 base.
- Provisioning of the Snom M700/M900 multi-cell DECT system.
- The call history now also shows calls that were transferred to you via several levels (previously only the first level).
- The dial status (effect on the "mode" in the dial plan steps) is now more consistent in connection with PUSH than before (especially important because of logon/logoff of the token).
- Mitel telephones are now provisioned to show missed calls.
Linked software versions:
wwphone 3.0.24 (Win, Linux and Mac)
- RTX provisioning via wwgate now works with the new transmitters.
- If an announcement is played in parallel to the ringing, the system switches back to the correct on-hold music.
- A change of the CDRs in Asterisk 13.29 caused calls transferred via CTI to appear a bit strange in the call log in the current version. As a result, transferred calls were no longer visible in the user call history unless they had "subscribed" to the number from which the call came in. Now it is not only the first call recipient who sees the call in the call history.
- Swisscom has started to send the call signalling as early media for calls to mobile phones. By default, Asterisk stops sending early media if more than one channel sends early media on a parallel call. (This is understandable, since the PBX cannot know which channel is "important" and which is not). Now, if a parallel call was made involving ONLY targets (and more than one at a time) that were signalling via early media, you would not hear any call signalling, but would hear something as soon as the call was answered. As soon as there was a target that sent classic "Ringing" without SDP, everything was ok, with only one target with or without early media as well. Now 183 Session Progress with SDP is converted into a 180 Ringing towards the caller in the above constellation (parallel call, multiple participants, only participants with early media).
- php 7.3.11
- Presence server: in CTI mode, the sending of DTMF tones via CTI client sometimes did not work correctly (the tones had to be sent via telephone keypad instead).
- TAPI driver extensions:
Supports new call acceptance (if the terminal supports Event:talk).
Transfer with query.
Hold/Unhold (if the terminal supports Event:talk and Event:hold).
- Provisioning Gigaset N870 Multicell DECT.
- The admin can now see, upload and delete users' avatars under Users/telephones.
- Presence status changes are logged in a temporary table (as a record to show the customer if the system should have rung but the affected users were on Unavailable during the period).
- Activation and deactivation of the special circuits are also logged in a temporary table.
- The logins and logouts in the groups can also be displayed in the web interface, furthermore it is now also logged in the auto logout case.
- Under Locations, the assigned DHCP leases are now displayed if a VPN box (wwgate) is used.
- The Snom models D315, D385, 745, D712, D735, D785, D120, D305 and D345 should now be provisionable (only D385 was tested!).
- The LDAP address book is now also provisioned on SNOM devices.
- The configurable value for the admin password is now also taken into account during SNOM provisioning (previously, the default value was always provisioned).
- Provisioning Gigaset Maxwell 4.
- Provisioning Gigaset N870 Multicell DECT (please refer to the wiki).
- Certain loop scenarios were made impossible (call forwarding to a terminal in a call group in which the diverting user is a member again).
- The ping statistics now show whether or not it is a TLS connection.
- Due to the increased space requirements of compiling with gcc 8.3, snmp is now removed from the distribution during booting, unless you set the "keep_snmp" flag in the config_table (rarely used so far).
- New firmware versions for Snom (depending on model) and Gigaset (3.13.13) via autoprovisioning.
- Now it is possible to configure for which numbers a user should see the call log even without an activated CTI licence (important for the end user web portal and for e-mail notifications for missed calls).
- asterisk 13.28.1
- php 7.3.9
- The LDAP settings were not fully provisioned to the Mitel terminals.
- LINK with additional (not visible) participants in the dial plan now works correctly.
- Android Push: Migration from gcm to fcm.
- Android Push: if Google's API does not respond, another send attempt message 3 seconds later.
- Android Push: a new blank message is sent to Android devices every 12 minutes. Reason: Sunrise closes inactive TCP sockets after 20 minutes. However, Google only sends keepalive messages every 28 minutes. This is actually a Sunrise or Google bug (depending on how you look at it) and affects all apps that use the Google GCM/FCM API. Now, if someone has not received a push notification (from any app) for more than 20 minutes and is a Sunrise customer, they may not receive any push notifications for about 10 minutes unless they are connected to the WLAN. More about this also under: https://eladnava.com/google-cloud-messaging-extremely-unreliable/ and Google has refused to lower the 28 minute interval for years....
- Extended logs for push notifications (when was what sent, did the API return an error?)
- Downgrade of the Yealink firmware from 84.0.10 to 83.0.50. Reason: the 84 firmware does not support DTMF in early media scenarios!
- Linking of the current versions of the clients (3.0.9).
- Penalty in on-hold queues was not displayed correctly in the web interface since KVM204
Mini optimizations (update optional):
- The subject for the VoIP SSL certificates is now also on the IP and the internal IP (only essential in VPN configurations with "auto packet diversion", so primarily a few wwcom customers).
- On the RTX multi-cell DECT system, only the G711a codec is now activated (G722 massively reduces the maximum number of possible parallel calls).
- The RTX multi-cell DECT system is now provisioned so that the system can control which ringtone is used on specific numbers (using the ringer field in the dial plan table, as with other devices).
- Linking of the firmware 02.101 for the Panasonic UDS124 radio base.
- Linking of wwphone/CTI 3.0.3 for Linux and 3.0.5 for Windows.
- Now TLS can be switched on and off per user for wwphone & wwmobile.
- In the ping statistics, all previous versions of KVM20x (for devices that were already offline) still showed values from the previous day that seemed to be from the current day (e.g. statistics from 15:30-16:30 at 16:30). This has been fixed. However, e.g. a connection from the previous day between 15:45 - 16:30 or similar is still displayed at 15:30 (correctly and deliberately) (since entries listed in the future obviously originate from the previous day!).
- The internal use of the diversion header set by a terminal can now be disabled (flag default_send_diversion in the config_table). The previous behaviour was kept, the new behaviour only occurs if you set this flag to 0 (default is 1). Disadvantage when deactivating: if you set up call forwarding directly on the terminal, you will not see on the target device that it is a redirection. Advantage: in certain scenarios the AareSwitch sends a Diversion header on the trunk. Since a Diversion header is then already set, Asterisk does not allow overwriting. This means that the Diversion header cannot be used in scenarios where a Diversion header already exists to display the dialled destination number with label on the terminal. Possibly wwcom will modify the source code of Asterisk for upcoming versions to allow overwriting, which could make this flag ignored again in the long run.
- Asterisk 13.25 contains a bug that causes pjsip show contacts to show all users twice. This resulted in twice as much data being stored for the ping statistics as needed. This meant that /tmp/temp.db needed twice as much memory as necessary. The duplicate entries are now ignored.
KVM204v2 is now available; these are minor customisations, so updating systems with KVM204 is not necessary (unless one of the customisations is explicitly needed).
- It now checks every 6 minutes instead of every 15 minutes to see if old traces need to be deleted.
- RTX 8663_v0410_b0012.fwu firmware is provided to ease the update process (required intermediate version if you want to upgrade to versions > v420.).
- Linking of wwphone/CTI 3.0.2 for Windows.
- In Flex Client mode, it is now also indicated when secondary audio devices are in use (assuming the Flex Core version is also >=3.0.2).
- When switching to CTI mode, the info that the secondary audio device is used was not hidden.
- If a MAC address was changed in provisioning, the dummy profiles for user mobility (getting a different config on the phone via *000) were not immediately recreated, so user mobility for the new telephone only worked after the config for Asterisk was recreated triggered by other config adjustments on the PBX.
- When renaming a PBX (hostname and/or domain), the Asterisk TLS certificate was not recreated (and thus applied to the old/wrong domain).
- Support of TLS/SRTP in cooperation with wwphone/CTI from version 2.3.99 (Attention: in the PBX settings, SRTP and TLS must be activated first. Reason: because other firewall ports are used (TCP 5061), wwphone should not stop working after an update => the IT specialist should therefore have the possibility to plan the changeover). However, under certain circumstances, it would make sense to activate TLS and SRTP by default for new connections => as this is only intended for wwphone/CTI/wwmobile, SRTP is only activated for users with a CTI licence!
- RTX 8660 and 8663 provisioning
- Aastra/Mitel 67XXi and 68xxi provisioning
- Gigaset N510 and Maxwell 3/Basic provisioning
- Snom D745 provisioning
- Supervisor function (CTI licence required by the supervisor): the supervisor can listen in on employee calls. For this purpose, he can establish the necessary connection in the CTI client by right-clicking on a user with an active call (listen-in mode: he hears everything that the customer and the employee say. Coaching mode: he can not only listen in, but also talk to the employee without the customer hearing him).
- Linking firmware 84.0.10 for Yealink T4xS models and 83.0.50 for T4xG models.
- In the on-hold queue, agents can be configured to be logged out automatically if they do not answer a call by the timeout. The agent is then no longer controlled until it proactively logs back into the group. This function is especially useful in connection with "Free for the longest time" and similar distribution strategies.
- Now lighttpd (without encryption) also runs on port 8099 => this is intended for scenarios where you work with phones that cannot be provisioned cleanly over https. Attention: the complete data transmission runs without encryption and is therefore sensitive and to be used at your own risk. The problem: e.g. Cisco ATA boxes refuse to communicate with a wildcard certificate...
- In the PBX settings, an API key can now be stored for tel.search.ch (since tel.search.ch is now much more restrictive with the number of queries per IP) => with the key you can still make 1000 queries per month, the customer must get the key themselves! One-time key per customer...
- In the past, special circuits were automatically activated and deactivated by all users in always mode immediately after they were created and applied to all numbers. If an admin clicked on "New special circuit" and forgot to configure the special circuit, it could happen that an employee overrode the dial plans of all telephone numbers by activating the special circuit. The special circuit is now created in such a way that no one has access to it and that it does not apply to any number.
- Downloaded faxes no longer contain the prefix "LumiMagic-" in the file name.
- The menu is now scrollable, if higher than the browser window.
- Linked wwphone/CTI version: Mac: 188.8.131.52, Linux: 2.3.15, 184.108.40.206
- Announcements were not overwritten when recording via telephone if the recording was started via web interface and the name was set the same as for an already existing announcement. (Overwriting when uploading and generating an announcement, on the other hand, worked correctly).
- Cleaning_cdrs script has been adjusted: if /mnt/history.db already exists but does not contain a database schema (can happen if you look at the call history without having done the rotation once before), the database will be created correctly.