Completely rewritten code for asterisk multi server management. The old code uses a series of SIP Header to transmit information about a call setting from a server to the other. This approach shows his limits. The number of information to transmit raises every day (music on hold, languages, etc. etc) and there is an upper limit on the size of the UDP packet. This new way uses still a SIP Header but in a packet way to save on precious UDP bytes. Obviously the risk of some side effects ( regression bug) in this rewrite is high, so please upgrade only during the weekend so we can fix any error without big disruption of service</dd><dd>Fixed a problem in the queue editing, when replacing a member with the same member but with Additional Destinations set</dd><dd>Improved CSV import function to allow importing also CR terminated files (Mac)</dd><dd>Fixed notify missed call notification language selection.</dd><dd>Improved handling of authentication data from Mail to fax ... I was decoding for new lines, but if the data entered was not ending with a new line, the last row was not used as authentication</dd><dd>Increased support for date decoding in voicemail to email... now it is able to decode better US locale dates</dd><dd>Added remote log activation/deactivation for devstatesender/receiver using an AMI User Event</dd><dd>Added a new status icon for devstatesender and receiver check</dd><dd>Added a new wallboard element to show average hold time
