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Firmware Version:  GST-1.01-72
Date:  2017-7-12

ChangeLog:
    1. "Forwarding to VoIP Number" suports SIP URI now (Example -- 101@192.168.2.2:6060)
    2. Add a option "Channel Number" in SMPP SMSC, support to overwrite "Destination address" with SIM Number of channel
    3. Optimized the RETRY process while sending Long SMS
    4. Optimized the RETRY process while GSM connection logout
    5. Optimized the Get_Number by USSD method

Firmware Version:  GST-1.01-71 
Date:  2017-5-27

ChangeLog:
    1.  Added “Call Count Limit” in SIM/UIM settings
    2.  Added “Module Registration when Limit expires” option. During a channel is deactivated for making calls, If this option is enabled (default setting), the channel maintains it
         sregistration to the cellular network.  Otherwise, the GSM module of the channel is shut down until the channel becomes active again.
    3.  Expanded the dialing rule to include the “+”symbol.
    4.  Changed local SIP port from fixed mode to random mode for SIP registration (this does not apply to Trunk Gateway mode).
    5.  Enhanced the connection stability for Remote SIM operation especially when sending bulk SMS.
    6.  In VoIP Advanced Settings, added “Line Selection Mode” which can be set to one of the 3 choices below. 
           a)    The least number of calls
           b)    The least amount of talk time
           c)    One by one in circular sequence (Round robin)
    7.  Corrected a number of minor bugs.

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