Release Notes 2.0.3
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SRTP
The rollover counter was not compatible with the popular library for SRTP. This needed to be fixed. This is a critical change. If you are using SRTP, you should consult your phone vendor if the rollover counter works correctly with the PBX and/or test secure calls with a duration of at least 23 minutes. The PBX does not check any more for old bugs, and uses only the correct way of calculating the rollover counter.
T.38
It seems that the T.38 passthrough got broken in several installations during the upgrade from 1.5 to 2.0. Now this seems to be fixed, at least in all known cases.
Agent Groups
The agent group may now send a instant message (IM) to extensions whenever the queue status changes. This way, agents can see how many calls are in the queue by looking at the display of their phone. This feature is specifically interesting for “low-profile” groups where a dedicated solution on the PC does not make sense.
The agent group call status window was showing also connected calls. This did not meet the expectation of customers; therefore the list now shows only the waiting calls.
Busy Lamps
There were additional situations when the status of the LED was not correct. One case was when a call was initiated through an auto attendant, which call was shown as connected, although the called extension was ringing. From a technical point of view this behavior could be seen as connect; however it did not meet customer’s expectations therefore we changed it.
Bandwidth Control
We added a feature that allows the limitation of bandwidth based o IP addresses. This way, the PBX can make sure that certain address ranges are not using too much bandwidth and tear the respective call down. However, due to the complexity of the problem we don’t promote this feature as a well-tested solution and customers should use this feature with care.
Caller-ID indication
When calls were redirected to cell phones, there was a problem with the indication of the original caller ID. We solved this problem by introducing a new variable in the explicit caller-ID indication of the trunk (“$f”). In this scope, we also changed the semantics of the “$a” flag to be used only when there is a tel-alias available. The default value for this setting is now always “$f $a <account>”, which should be a good solution in most cases. Customers can still overwrite this default.
Conferences
Conferences can now be setup by the user through the web interface. The PBX can send calendar invitations to the participants of the conference that include the PIN code for the conference. The moderator of the conference has now the possibility to close a conference by dialing a star code (“*9”).
Denial of Service (DoS) Protection
The PBX now limits the number of new calls per second. This way, relatively primitive DoS attacks are defended in a reasonable way.
Plug and Play
The plug and play configuration for the Polycom devices now contains a parameter that decides which transport layer should be used. This way, installers can decide o use TCP transport layer. This transport layer is necessary if relatively long packets (more than 1492 bytes) need to be sent to the Polycom devices, for example when the buddy lists are getting too long.
Click to Dial
The PBX generates links for the click to dial feature in several places, for example in emails. Those links were http in the previous version. In the new version, the PBX will choose https if there is at least one https port specified to the PBX. This way, the click to dial feature will not expose passwords to the network. However, because most web browsers complain about untrusted https certificates, this increases the need to load a valid certificate into the PBX.
Scheduler
In Linux, there were still problems with the scheduler. In the embedded versions, the scheduler simply did not work as specified in the Linux documentation. In order to avoid the problem, the PBX now calls sched_yield() periodically in uncritical threads, so that the jitter in the RTP play out should be practically gone. In the PC-based versions, the RTP play out thread now uses the SCHED_RR algorithm, which seems to solve the problem even under heavy load.
Star Codes
In many systems, users can go to the mailbox by dialing a number that is not a star code (e.g. 5500). We added this possibility also in the PBX. This feature is limited only to the go to mailbox-star code but can be extended in future also for other star codes.
The German prompts for DND on and off were mixed up.
CDR
There were still problems in the serialization of the SOAP requests for the CDR reporting. These should be fixed now.
Also, the PBX now also reports CDR for transferred calls. For this purpose, the PBX breaks the call up into two calls and reports them separately.
The CDR that are written to the file system did not contain the statistics information. This was a pity, because the statistics can be very useful for debugging purposes. Therefore, we included them. Now system administrators can use the CDR on the file system to see how many packets have been send and received during a call and how the jitter buffer fill status was.
External Calls
When the user entered the pound key without entering a destination number, the PBX was processing a empty destination. This was causing some confusion, especially when authenticating the call through an external application server. In the new version, the PBX does not accept the pound symbol in this context.
Mailbox
When moving a mailbox message to another user, the owner of the message can now add a comment to the message. Because we don’t have a IVR recording for this feature, the PBX currently does not explain this possibility.
Also, the PBX now immediately sends a MWI out when the user deletes a message.
Emails
The UTC offset in emails were not correct, this was leading to incorrect timestamps.
Support for QOP
Some service providers accept only QOP Digest authentication (for example, to authenticate ACK requests). This was added to the 2.0.3 build of the PBX.
BYE Reason
In certain situations, it is useful to know why the PBX disconnects a call. For this purpose, the IETF has defined the Reason header (RFC 4411). The PBX now uses this header when the BYE was sent because of bandwidth limitation or because of a RTP timeout. This should make troubleshooting easier.
Attended Transfer
In an attended transfer it was not clear which call "survived" the transfer. This was leading to billing problems, where the PBX was reporting the original call, not the transferred call. Also, attended transfer into one-legged calls (e.g. conference rooms) was not working.
NAT DMZ
There were several cases where customers requested to operate the PBX in a DMZ that does perform NAT. We added a simple replacement algorithm that changes the presentation of IP addresses to the outside world, so that this becomes possible. However, because of the difficulties in the setup, we still strongly recommend not to use NAT together with DMZ.
Gain Control Support
Finding the right gain was increasingly becoming a problem. The PBX now has a feature that allows to measure the energy of the calls legs measured in dB. This feature needs much more CPU, as all media must be decoded - however it is very useful to find the right gain values when using analog components like FXO gateways.
Windows 2000 Support
The PBX was using functions that were only available for Windows XP or higher. These functions increase the stability of the system. However, they were not available for Windows 2000 and this made it impossible to run the PBX on Windows version prior to Windows XP. The 2.0.3 version now uses the old functions if the operating system version is below Windows XP (Windows 5.1). The Windows XP functionality is not affected by this.
