Release Notes 3.3
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Intercom
When using intercom with more than one registration, it was random which phone would automatically answer the call. Because of this, we changed the behavior so that intercom only works when there is only one registration. Otherwise, the intercom will just be a regular phone call.
Callbacks
There were cases where the callback mechanism of the PBX could produce idle calls that are not deleted immediately. However, those calls were deleted after the maximum call duration.
Mailbox
There was a race condition between sending out emails and storing the voicemail on the file system. This race condition could make the PBX delete a message because it did not make it to the file system yet (see http://forum.pbxnsip.com/index.php?showtopic=2111).
In the new version the administrator can edit the text templates that are used for sending out emails. It is not necessary to edit the texts in the file system any more.
The naming of email in the spool directory was unfortunate. Emails could get stuck at the head of the queue, blocking all other outgoing emails. The new scheme automatically renames the emails and puts them at the end of the queue.
Emails for missed calls now contain images so that it becomes easier to add them the personal address book by a single click. Welcome emails can now be sent from the web interface. This is very useful if a user has forgotten the password; the administrator just has to click on the link to re-send the welcome email.
We added a new system email event for user-initiated disconnected during one-way audio. We have seen customers that have problems with the stability of their Internet connectivity. When the call stalls, users usually hang up after a few seconds while the one-way audio is still going on. In this case the PBX now sends a notification to the system administrator. The reporting of one-way audio disconnect was not enough because usually the signaling channel still works and both sides clear the call in a regular way.
Dial Plans
Dial plans can have a global scope now and the administrator can lock domain administrators out from editing dial plans. This makes it a lot easier to offer a self-service hosted PBX where customers can change accounts, but not dial plans. The dial plan name can now be changed (this useful feature was missing for a long time).
Trunks
When a trunks had no more CO-lines available, they would not failover. That was causing problems in deployments that want to balance the load on the trunks. Therefore, the new version supports failover on failure to allocate a CO-line. This failover works as if the trunk would have returned a class 5xx SIP response.
Each domain can now have a default ANI, so that it does not have to be specified for each extension or each trunk.
Calling card
It is now possible to dial automatically from the calling card. For North America, the PBX can now automatically start the call after 10 or 11 digits. This makes it easier and faster to use the service for end-users.
Caller-ID presentation
For redirected calls from a trunk to another trunk, there was a common problem with the presentation of the original caller-ID. Most service providers don’t trust the PBX when sending out the caller-ID. In order to prove that the caller-ID is valid, the PBX now shows the original Call-ID in the outbound call request. Service providers who are able to verify this hint can now send the original caller-ID.
Media processing
It is not possible to lock the codec that has been negotiated. This feature is useful in environments where connected devices or services cannot dynamically change the codec (although they advertize this ability). The lock can be set on a global level and also on a trunk level.
If seems there were cases where the fetching of audio files could stall or at least be dependent on other events. The relatively old code the fetching these files was updated.
There are still cases where the signaling of the RFC2833/4733-codec is not implemented correctly by connected devices or services. In order to avoid interop problems, the PBX now tunes automatically to the provided codec number. This should make the OOB DTMF codec detection very robust.
QoS
The PBX now can also set the DiffSrv bits of the SIP traffic. This was not the case before, SIP traffic was sent on the default traffic class. There was already a setting for the RTP traffic class, but the setting for the SIP traffic was missing. Also, the traffic class can now have a symbolic name, like "cs5".
Registration
The number of registrations can now be limited on a global scale, on domain level and on extension level. This is important because every registration can cause a fork of a call and might result in an over-usage of system resources. It also may be a problem when subscribers are abusing extensions which now can be regulated by the operator.
Call Recording
The Call-ID can now be part of the recording name ("L"). The new version also fixes a bug with the extension override of the recording preferences of the domain. Sometimes, the setting of the extension was ignored and the domain setting was used for making the decision if the call should be recorded.
Emergency Calling
In order to support the North American requirement for emergency call back, every extension automatically gets an EPID (end point identifier). This EPID is used when placing a call to an emergency call center, so that when the call center calls back the PBX routes the call directly to the extension, even if the call usually has to go through an auto attendant.
There is a new setting that makes it possible to record only calls to the emergency numbers. This feature is very useful for schools and other public installations where the emergency service must be available, but abuse should be minimized.
ACD
The ACD now makes it possible to remember the agent that received a call from a specific caller-ID. The next time when the same Caller-ID calls into the agent group, the PBX tries to connect the call to the last agent. This feature is still "beta".
Hot Desking
When using the hot desking in a multiple-domain environment, it could happen that the redirection went into the wrong domain. This problem should be fixed in the new version.
Shared Lines
When seizing a shared line for making an internal call, the PBX did not automatically release that line after the call started. That was a problem because the user has to release the line after the call manually or wait until the PBX would automatically release the line.
CDR
The PBX now includes the source IP address of the caller in the CDR for a call.
Plug and play
The plug and play mechanism was revised. Now there are essentially two ways to perform plug and play. The first one uses the MAC address; the second one uses HTTP authentication.
The MAC-address based does not take the MAC address out of the requested file; instead it takes it from the ARP cache of the host. This mechanism is easier and it makes it more difficult to steal identities in the network.
When using the HTTP authentication, the device must send the username and the password. This makes it safe to provision devices even in hosted scenarios.
The plug and play for snom phones and for Polycom phones has been updated to reflect the availability of updated software. It is now possible to specify multiple MAC addresses per extension to provision more than one phone per extension. This is very useful if a user has for example a desktop phone and a cordless phone.
The assignment of buttons can now be changed by the end user. If the administrator assigns a button profile to an extension, the extension user can now override the setting from the web portal. This is very useful to deal with requests like setting up personal speed dial codes on buttons.
The usage of buttons now also checks the proper permission to view the button. In previous versions it could happen that a user could subscribe for a button without having the permission to see the extension’s status. This bug was not very serious as only the domain administrator could assign button profiles.
The built-in NTP server now also supports IPv6.
The time zones for the snom M3 have been addressed and corrected.
The provisioning of day light savings for Polycom devices was buggy.
The provisioning now also respects the IP replacement list. That makes it possible to use the automatic generation of configuration information also in setups where the PBX is running in a DMZ with a seperate IP address (see http://forum.pbxnsip.com/index.php?showtopic=2164).
Midnight events
There is a new midnight event for resetting the blocking of caller-ID. Like with the other events, it should help to lower the number of support cases where users complain that they are not able to make outbound calls because of the policy of the carrier or they are simply showing up as "anonymous".
The midnight events now also respect the local time zone of the domain or the user, respectively. This helps giving users across different time zones a completely local impression of the PBX service.
Status screen
There is a new statistic in the status screen that shows the number of registrations and the number of registrations and subscriptions. This overview will help to localize sudden registration dropouts, for example because of a network failure. This registration overview is now also included in the midnight status email.
The status screen now also contains the current system time. This is useful to see if the time and time zone configuration works.
Routing settings
When using the routing settings on TCP or TLS transport layer, the PBX would not replace the local IP address with the IP address that comes from the routing setting.
When changing the routing settings, the PBX now flushes the routing cache, so that changes have immediate effect. In the previous version, administrators had to restart the service or wait up to one hour until the chance took effect.
CS410
The built-in PSTN gateway had problems when the carrier sent information after the call was disconnected. In the 3.3.1 release, this information is ignored.
