Firmware Version: GST1610-1.01-64-1 Date: 2018-07-26 ChangeLog: 1. Fixed the problem that part of channels can not start in small probability. 2. Optimized dial plan feature
Firmware Version: GST1610-1.01-64 Date: 2018-07-17 ChangeLog: 1. Enhanced compatibility for remote SIM feature 2. Fixed the bug about Base Station Poll mode 3. Fixed the problem about that the Channel2 will be skiped in "Line Selection Mode-Round Robin" mode 4. Fixed the problem about that the SMS Outbox can not save few characters, such as "@" 5. For safety's sake, forbid to downgrade firmware version after this firmware
Firmware Version: GST1610-1.01-63 Date: 2018-05-18 ChangeLog: 1. Adjusted some default settings (DNS and RTP Disconnect Detect) 2. Added "Toll-Free Prefix" feature. those calls that match this Prefix will not be recorded in CDR and not deduct Remain time 3. The "CID Prefix" is effective for each channel now 4. Optimized "Get Number" features, added a new method "SMS to Known Num" 5. Some other Optimizations
Firmware Version: GST1610-1.01-62-4 Date: 2018-01-19 ChangeLog: 1. Fixed the problem about sending SMS via SIP MESSAGE 2. Fixed the auto-reboot problem in certain specific situations 3. Limit the code range of "GSM-SIP Code Map", from 400 to 699
Firmware Version: GST1610-1.01-62-3 Date: 2017-12-25 ChangeLog: 1. Fixed the issue about that overmuch "486 Busy" causes low ASR 2. Fixed the issue about that "None Ringback Timeout" feature will be invalid after reboot
Firmware Version: GST1610-1.01-62-2 Date: 2017-12-21 ChangeLog: 1. Optimized SIP registration to make VoIP online faster, espacially after sipcli restarting 2. Fixed problems about "Call Waiting" 3. Fixed and optimized the mechanism of "Sending SMS via SIP Message Request 4. Changed the default setting of VAD to "Enable" 5. In "Remote SIM" mode, support to check "Include no connected call" for Limit Call Counts by Period (Require v1.9 of SIM SERVER or newer)
Firmware Version: GST1610-1.01-62-1 Date: 2017-11-27 ChangeLog: 1. Optimized the process when sending concatenated SMS failed, not to continue sending the remaining sub-SMS if foregoing SMS send failed 2. Added "Auto Config" button on "SIM Forward, GSM Carrier, GSM Base Station" pages 3. Optimized the mechanism of releasing SIP transaction 4. Reformed "When X calls occurs, Deactivate the line for Y seconds" this feaure to be as "Deactivate the line for A-B seconds" 5. Limited the "Black/White List" of Call Auth" as 999 bytes 6. Remember the checked Line ID on "Send SMS" page when go "Back" after sending SMS
Firmware Version: GST1610-1.01-62 Date: 2017-10-24 ChangeLog: 1. Optimized SMS-Forwarding Features as following: a) Support to forward SMS to VoIP in "Trunk Gateway" mode b) Forward the concatenated long SMS after combined c) Add the Channel-ID while forwarding to mobile d) Add the Sender Number while forwarding to email e) Adjust SMS-Forwarding Features to be effective for each Channel 2. Support to UP/DOWN modules by multiple choices on the status page 3. Fixed the problem that "None Ringback Timeout(s)" feature got invalid 4. Reformed the "Call Out/In Auth" features 5. Optimized CDR statistics about Call-Counts and Duration 6. Added a feature about "consecutive fast-answered calls" in Event Triggers 7. Some other Optimizations
Firmware Version: GST1610-1.01-61-1 Date: 2017-9-21 ChangeLog: 1. Enhanced secureity capacity 2. Fixed the problem that Line2&3 of GoIP16 (Seriel Number begin with "16MCTRM", since May-2017) sporadically restart by itself 3. Added "Dynamic Concurrence Capabilities" item in Advance VoIP page, it is used to send GoIP's current quantity of available Lines to SoftSwitch Server (latest VOS supports)
Firmware Version: GST1610-1.01-61 Date: 2017-8-24 ChangeLog: 1. Sloved the problem that part of channels' VOIP status occasionally goes offline while "Trunk Gateway" mode 2. For SMPP appication, increase the life time of SMS delivery report to 10 hours 3. Add "Get Callee Number" feature in Advance VoIP page, choose to get callee number "By Request Line" or "By To Header" 4. Adjust the mechanism of telephonometry, make it to be more accurate 5. Add "Talk Time Adjust(s)/Call" feaure in SIM page, to add 1 second for all outgoing calls (Generally, the Provider's billing system will delay 1~2 second for hangup) 6. Add "GSM None Ringback Timeout(s)" in "Call Out" page. Hangup the call if it still isn't RingBackTone beyond the specified seconds after dialing 7. Fixed the problem that Current time incorrectly shows year 2036 8. Fixed the problem that GSM module abnormally start while it is being closed
Firmware Version: GST1610-1.01-60-2 Date: 2017-7-12 ChangeLog: 1. "Forwarding to VoIP Number" suports SIP URI now (Example -- firstname.lastname@example.org:6060) 2. Add a option "Channel Number" in SMPP SMSC, support to overwrite "Destination address" with SIM Number of channel 3. Fixed the ERROR while SMPP SMSC deliver a long SMS greater than 255 bytes 4. Optimized MC8618 module of CoIP 5. Optimized the RETRY process while sending Long SMS 6. Optimized the RETRY process while GSM/WCDMA/CDMA connection logout 7. Optimized the Get_Number by USSD method 8. Updated the DSP driver of GoIP4 to fix voice fault
Firmware Version: GST1610-1.01-60-1 Date: 2017-6-26 ChangeLog: 1. Optimized SMS receiving, concatenated long SMS can be combined now. 2. Add VAD function, not to transmit media in silence time, this can save lots of bandwidth. 3. Web page of device supports to send long SMS now (up to 1024 bytes ). 4. Corrected a number of minor bugs.
Firmware Version: GST1610-1.01-60 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. Increased the volume output of the GSM module (M35) to the cell network. 5. Changed local SIP port from fixed mode to random mode for SIP registration (this does not apply to Trunk Gateway mode). 6. Enhanced the connection stability for Remote SIM operation especially when sending bulk SMS. 7. 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) 8. If a SIM require PIN authentication, disable the PIN authentication after the success of authentication. Then the SIM no longer need PIN code. 9. Corrected a number of minor bugs.