Grandstream HT502 ATA Firmware 1.0.10.9

Manufacturer:

Description

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Fixes:

- Fixed BroadSoft interop: device cannot switch between two lines after receiving server NOTIFY
- Fixed Voice Frame per TX missing from Data Model
- Fixed [TR-069]: CPE doesn't report gateway related info in the TR-069 INFORM
- Improved message receiving reassembling logic for SIP over TCP/TLS
- Fixed fail to parse CID when receiving distinctive ring tone dr2/3/4/5/7
- Fixed HT502 FXS1: connect analog phone and network cable to HT502, reboot by IVR, when DUT start, FXS 1 may become unavailable
- Fixed event lost during performance testing
- Fixed cannot split off the call
- Fixed LEC is not disabled properly
- Fixed cannot change Use Configured IP without reboot. Fixed the problem when the device does not follow record route header route, resetting host to NULL causes device to use IP provided by DNS in configIP mode.
- Fixed ringing does not timeout
- Fixed TLS block read issue where the read may not return the complete message
- Fixed device does not send RTCP BYE message
- Fixed device does not follow record route header route
- Fixed malformed "To" & "Request-URI" when using this dial plan
- Fixed wrong order on Route header with ACK and BYE requests
- Fixed Broadsoft Interop: device does not update "anonymous" with "Privacy:id". When there is no PAI header but there is a "Privacy:id" header, we need to check if CID exists in To: header, then replace CID with "anonymous"
- Fixed Broadsoft Interop: Connected Line ID is not updated after Call Transfer or Call Forward.
- Fixed problem where Connect Line ID was not updated when we receive the 2nd 18x in a backto- back 183-180, 183-183, or 180-183 case
- Fixed crash problem when "subscribe MWI" is changed from "No" to "Yes"
- Fixed an issue after receiving inbound Caller ID with ";" semicolon
- Fixed Payload Type set to 101 have no audio in the call
- Fixed HT503 FXO port will pick up even without PSTN line connected
- Changed year 2012 to 2013 on web UI
- Updated SRTP library to fix issue of processing odd bytes of RTP payload
- Fixed a Current Disconnect problem
- Fixed a Polarity Reversal problem
- Fixed FXS2 port wrong default value for "Preferred DTMF method"
- Fixed some Time Zone settings cannot be selected from Web UI
- Fixed device still tries to send REGISTER requests to Failover SIP Server even though it has successfully registered to Primary SIP Server
- Fixed if Firmware Server lost network connection during firmware upgrade device will not work properly
- Fixed device cannot hear the other party after receiving ACK with no SDP
- Fixed device loses Dial Tone after power cycling unit once after factory reset
- Fixed a memory leak problem when SUBSCRIBE for MWI is set to Yes
- Fixed Prompt Tone switch slow and with wrong frequency
- Fixed device crashes during Direct IP Call under certain conditions
- Fixed Broadsoft Interop: device Connected Line ID is not updated after Call Transfer or Call Forward if received 183
- Fixed Call Waiting tone heard when there is no Call Waiting call
- Fixed when dial plan is set as call features do not work

New Features:

- Added Minimum Delay Before Dial option
- Added support for receiving TCP/TLS "\r\n" keep-alive messages
- Added support for call routing based on SIP request URI user ID
- Added support for Telkom SA special 3WC method
- Added SUBSCRIBE messages to multicast address from UCM61xx
- Added support for sip NOTIFY 'resync' event
- DialPlan Enhancement: modify "." to indicate there is zero or more of the preceding element
- Added support for compact session timer header
- Added support for HT502 Blacklist option
- Added support for new device LED pattern. Added new option in HT502 Advanced Settings so that user can select LED Pattern A (original) or B (new)
- Allowed HT503 to set Hook Flash Duration to 200ms




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