3CX Phone System 7.1 RC2 released

We are pleased to announce 3CX Phone System 7.1 RC2 – it is pretty much good to go.

In this build we have made a big improvement to the Tunnel service. If UDP is available, then this will be used, otherwise TCP will be used. On low bandwidth connections this should result in clear audio improvements. Furthermore the reconnection mechanism has been improved, resulting in better reliability. The VOIP phone was updated accordingly.

Also, we have made some tweaks to the Cassini web server option and this is now also recommended. The interface runs a little slower on it, but its completely separate from other IIS applications, as well as IIS updates, so Cassini does have its ‘charm’.

The fax sending module was updated and now works well with 7.1

Soon a small update to the 3CX Assistant will follow with some minor improvements.  This will not require a server update. In the pipeline for release over the next few weeks is our Outlook and Salesforce.com integration, which will be a significant step forward on the old voip client outlook integration.

Included also is a SKYPE gateway. This is pretty big news however this has been eclipsed somewhat by the recent announcement that SKYPE will have a SIP interface in the very near future. The 3CX SKYPE gateway is a re-distribution of the GPL SIPtoSYS gateway by mhspot. Details and download here. If you are not in a hurry for SKYPE integration, i suggest registering for the SKYPE sip beta at http://www.skypeforsip.com/ as this will be the officially supported interface by SKYPE.

The full change log:

  • Added: Much improved tunnel that can also support UDP (if available) for better audio quality
  • Added: Skype Gateway – 3CX Gateway for SKYPE
  • Added: Improved Cassini support – Now a recommended web server
  • Added: Sangoma A200 FXO PCI card (BETA – North America only)
  • Added: Portech MV372 GSM gateway template
  • Added: Actio.pl VoIP provider (Poland) template
  • Added check for deleting of the operator extension. Extension cannot be removed unless modified to something else.
  • Added: Added the ability to restore Call History logs.
  • Added: Detailed backup and restore logs during database operations
  • Fixed: Polish Prompts
  • Fixed: RTP Port leaks in tunnel
  • Fixed: Exception on 2 Slave configuration
  • Fixed: Faster reconnect  of tunnel when connection is lost.
  • Fixed: Source ID reorganization and changes to the Database.  Easier
  • to Add Source ID rules now./  Removed the need to type them in twice.
  • Fixed: Invalid Time Interval in “In Office Hour” Selection.  00:00 is invalid as to range.
  • Fixed: Exception in some rare configuration instances
  • Fixed: In some instances a thread would not stop in the wizard
  • Fixed: Removed ability to enter a blank source identification value.
  • Fixed: Restore of a file without an extension (provisioning) was failing.
  • Fixed: IVR redirection when voice mail is disabled. When VM is disabled you get the correct prompt
  • Fixed: Importing of new calls after Call History Import in Call Reporter
  • Fixed: Interpretation of Make call calls in the Call reporter
  • Fixed: interpretation of Sip Forked ID extensions.
  • Fax server: Fixed Nonce authentication expiration.
  • Fax server: Added UAC compliancy
  • Fax Server: Added support for Domain name resolution
 

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  1. JohnCz

    Great turnaround time..thanks. I’m not interested in supporting Skype at the moment but the SkypeForSip is an interesting development. I checked the domain, and it doesn’t seem to be owned by Skype…are you sure this is legit?

    Also, I respect you include an alternative but I think its a mistake not to default to IIS. Perhaps you can detect if IIS service is running. If not, default to Cassini…if it is running, default to IIS.

    March 25, 2009 at 10:57 pm
  2. The link for the fax server downloads the installer for the VOIP phone!!

    March 25, 2009 at 11:21 pm
  3. the fax server link downloads the voip phone installer.. .

    March 26, 2009 at 1:10 am
  4. Justin – fax server link fixed… Thanks for letting me know

    March 26, 2009 at 12:50 pm
  5. John,

    THe user gets offered the option of IIS or Cassini, its up to the user to decide…

    March 26, 2009 at 12:51 pm
  6. John Wigley

    Can I ask if there’s any technical explanation of the new tunnel functionality anywhere ? One of the things I think you could make a real impact with, is to support multiplexing multiple RTP streams into the same packet for bandwidth reduction over constrained ADSL lines, thus allowing them to support a far greater number of concurrent calls. No SIP PBX supports this as far as I’m aware, although Asterisk has done it natively for many years with IAX2.

    March 31, 2009 at 6:03 am
  7. JohnCz

    I would use the following Skype own’d site if you are interested in their Skype to SIP program. Frankly, I’d be more interested in Skype if they offered monthly subscription rates.

    http://www.skype.com/business/form/sip-beta/

    April 1, 2009 at 1:15 am