Model Name: | KX-TGP600 CE/UK/C/AL/LA/LC/BR/RU |
---|---|
Existing version: | 2.104 |
New version: | 2.200 |
No. | Fixed issue |
---|---|
1-1 |
Improvement The status of SRTP call is displayed on LCD. New configuration parameter: DISPLAY_SRTP_CALL_ENABLE |
No. | Fixed issue |
---|---|
2-1 |
Modification Fixed issue: Spelling mistake of CANCEL_OPERATION_MODE parameter as CANCEL_OPERATON_MODE. New configuration parameter: CANCEL_OPERATION_MODE |
2-2 |
Modification Fixed issue: The device cannot send DTMF after receiving 183 while outgoing call. New configuration parameter: SIP_183_TALK_ENABLE |
2-3 |
Modification Fixed issue: The device cannot work correctly when CANCEL_OPERATON_MODE is set as 1. |
2-4 |
Modification Fixed issue: The caller's LCD does not show CID correctly, the end of the name B is visible. Operation and conditions are as following. Operation: B has CFNAnswer to C activated. A calls B. B rings and doesn't answer. C rings and answers after 5 seconds. Condition: The name of B is longer than the name of C. [ Handset Firmware Version ] -TPA60: Ver.02.04.007 -TPA65: Ver.02.04.009 -UDT121: Ver.06.06.009 -UDT131: Ver.06.06.009 |
2-5 |
Modification Fixed issue: The Safari browser does not show correctly. |
2-6 |
Modification Fixed issue: The device replies 488 when receiving INVITE whose SDP has 2 m line. |
2-7 |
Modification Fixed issue: The device replies 491 instead of 500 when receiving re-INVITE before receiving ACK. |
2-8 |
Modification Fixed issue: When using Bluetooth Headset, the voice path is connected to ear speaker after hold retrieve. [ Handset Firmware Version ] -TPA60: Not released -TPA65: Not released -UDT121: Ver.06.06.009 -UDT131: Ver.06.06.009 |
No. | Fixed issue |
---|---|
3-1 |
TGP600: Ver.02.104 -> Ver.02.200 [Handset firmware] TPA60: Ver.02.04.006 -> Ver.02.04.007 TPA65: Ver.02.04.008 -> Ver.02.04.009 UDT121: Ver.06.06.008 -> Ver.06.07.000 UDT131: Ver.06.06.008 -> Ver.06.07.000 |
Model Name: | KX-TGP600 CE/UK/C/AL/LA/LC/BR/RU |
---|---|
Existing version: | 1.177 |
New version: | 2.104 |
No. | Fixed issue |
---|---|
1-1 |
Improvement Personal XML phonebook each cordless handset by sending IPEI on phonebook search. New configuration parameter: XMLAPP_SELECT_HS_PB |
1-2 |
Improvement Support Hotel phone specification New configuration parameter: DISCLOSE_FUNCTION_ENABLE VM_FUNCTION_ENABLE SOFT_KEY_QUICK_DIALx SOFT_KEY_LABELx |
1-3 |
Improvement Cease to restart main unit every time uaCSTA becomes enable on wireless handset. Related configuration parameter: UACSTA_ENABLE_n |
1-4 |
Improvement SIP-TLS is once disconnected when some parameters are changed. |
1-5 |
Improvement Setting of 'Range Alarm' can be changed with configuration parameter. New configuration parameter: RANGE_ALARM_ENABLE |
1-6 |
Improvement Features to call forward or conference call hard to find and menu is not clear like TGP500 |
1-7 |
Improvement Located Embedded Web menu under SYSTEM SETTING because it's hard for a user to find it. Assignment under HANDSET SETTING remains. |
1-8 |
Improvement In dial plan input, allowed NULL replacement format, such as "<#:>". |
1-9 |
Improvement When a phone receives a call while a user is calling to the other party, the user needs to press CANCEL key twice to cancel the calling call. Improved the times of pressing CANCEL key to once. New configuration parameter: CANCEL_OPERATON_MODE |
1-10 |
Improvement Application log in TGP/HDV can be sent to SYSLOG server. (When wireless handset can't be registered in TGP600, the result can be logged.) New configuration parameter: SYSLOG_OUT_START |
1-11 |
Improvement Support Safari web browser. |
1-12 |
Improvement Operational value can be sent to SYSLOG server by pushing special code from handset. New configuration parameter: SYSLOG_OUT_START |
1-13 |
Improvement Supported Packet Sniffer function |
1-14 |
Improvement Supported Remote Interface of Wireshark New configuration parameter: PCAP_ENABLE PCAP_REMOTE_ID PCAP_REMOTE_PASS PCAP_REMOTE_PORT |
1-15 |
Improvement A user can confirm handset that administrator authentication is assigned on wireless handset. |
1-16 |
Improvement A user can confirm telephone number on line by easy operation, holding down 'Down' key. |
1-17 |
Improvement Off-hook monitor can be configurable to disable. New configuration parameter: OFF_HOOK_MONITOR_ENABLE |
1-18 |
Improvement Group Paging of BroadWorks. Ability to specify how to handle the 1st active call when phone has the call and receives INVITE with answer-after=0. New parameter: SIGNALING_AUTO_ANS_ENABLE_n |
1-19 |
Improvement Phonebook is downloaded with HTTP GET when device boots up. New configuration parameter: XML_PHONEBOOK_URL XML_PHONEBOOK_CYCLIC_INTVL |
1-20 |
Improvement Device starts ringing with delay of designated time when incoming call arrives. New configuration parameter: DELAY_RING_TIME_n |
1-21 |
Improvement The rest of participants can continue the call if organizer terminates 3 way conference. New configuration parameter: CONF_OWNER_OUT_ENABLE |
1-22 |
Improvement Support Private hold feature for Shared Line Appearance of Metaswitch. New configuration parameter: PRIVATE_HOLD_ENABLE |
1-23 |
Improvement Support "No touch provisioning" of Broadsoft |
1-24 |
Improvement Support "No touch provisioning" of Broadsoft |
1-25 |
Improvement Support character input timeout New configuration parameter: AUTO_INPUT_KEY_TIME |
1-26 |
Improvement Support Ring Splash |
1-27 |
Improvement Panasonic RootCA is embedded in device. |
1-28 |
Improvement We modified not to stop call following case. 1) Send Invite and receive 407. 2) Send Invite and receive 302 3) Send Invite and receive 407 |
No. | Fixed issue |
---|---|
2-1 |
Modification Fixed issue: When a phone receives a call, call information in PAI header' is displayed even though CNIP_FROM_ENABLE=�Y�. |
2-2 |
Modification Fixed issue: DTMF can be heard when configuration of DTMF_METHOD_n is RFC2833 or SIP INFO. New configuration parameter: DTMF_OUT_ENABLE |
2-3 |
Modification Fixed issue: Modify to work correctly when the dial plan includes Macro timer and DIAL_PLAN_NOT_MATCH_ENABLE is set as Y. |
2-4 |
Modification Fixed issue: Line Status of TR-069 isn't displayed correctly New configuration parameter: TR069_REGISTERING TR069_REGISTERED |
2-5 |
Modification Fixed issue: Result of Traceroute and Ping of TR-069 isn't correct. |
2-6 |
Modification Fixed issue: The handset receiving Multicast Paging can not hand over to the repeater. |
2-7 |
Modification Fixed issue: TGP600 does not boot up by updating FW again during update with Web. |
2-8 |
Modification Fixed issue: There is a case that the some data are missing when the phone sends syslog to the server. |
2-9 |
Modification Fixed issue: In spite of setting FWD busy, the phone doesn't transfers the incoming call when the session is full. We modified that the phone transfers the incoming call while establishing call when setting FWD busy. |
2-10 |
Modification Fixed issue: When setting shared call as enable under Metaswitch environment, the phone fails to attempt outgoing call. |
2-11 |
Modification Fixed issue: When a phone receives a call, it doesn't send '180 RINGING'. |
2-12 |
Modification Fixed issue: When you catch a call to shared number on a phone, the other phones continue to ring. |
2-13 |
Modification Fixed issue: You can't hear any voice (No RTP) in call of shared number. |
2-14 |
Modification Fixed issue: BLF doesn't work correctly after provisioning though it works after rebooting. |
2-15 |
Modification Fixed issue: When you press "System Setting" menu, a phone freezes while displaying "Please Wait". |
2-16 |
Modification Fixed issue: When using XML phonebook, "?" isn't attached after phonebook URL. |
2-17 |
Modification Fixed issue: The operation of HDV300 becomes unstable. |
2-18 |
Modification Fixed issue: DND setting is not synchronous to server after failover. |
2-19 |
Modification Fixed issue: Transferred call is terminated with blind transfer of incoming call from non-SIP device. |
2-20 |
Modification Fixed issue: Call is terminated after about 25 sec from starting conversation. (Server terminates call because device only sends "100 Trying" to "reINVITE". ) |
2-21 |
Modification Fixed issue: Server authentication is failed when server uses SSL re-session. |
2-22 |
Modification Fixed issue: Device doesn't originate by dialing target dial set with repeat of dial plan. |
2-23 |
Modification Fixed issue: "Forward No Answer" doesn't act. |
2-24 |
Modification Fixed issue: Device can't terminate incoming call when picking up call waiting during call with the device doesn�t' have call hold capability. |
2-25 |
Modification Fixed issue: Device can't return to standby status by pressing "Cancel" Softkey when hearing busy tone with busy of transfer target. |
2-26 |
Modification Fixed issue: Incoming LED doesn't go off with flashing when answering against recall of hold call while talking to outside call. |
2-27 |
Modification Fixed issue: Opposite party executes call hold and call release after CallPickup. Then call is terminated with transfer side answers against call transferred automatically. |
2-28 |
Modification Fixed issue: REGSTER isn't send when SIP related parameters becomes available with SPV of TR-069. |
2-29 |
Modification Fixed issue: Call is terminated after 25sec from beginnings of call. |
2-30 |
Modification Fixed issue: CallWaiting doesn't arrive when executing on-hook after opening LineStatus screen during CallWaiting arrives. Also original call can't be terminated. |
2-31 |
Modification Fixed issue: Pop noise occurs when changing protocol from RTP to SRTP. |
2-32 |
Modification Fixed issue: Response is slow for BLF settings of MetaSwitch server. |
2-33 |
Modification Fixed issue: To faster response when pick up a call with using wideband (temporarily �fixed� when using narrow band the delay acceptable now) |
2-34 |
Modification Fixed issue: MCP sending from the handset could not stop fairly infrequently. |
No. | Fixed issue |
---|---|
3-1 |
TGP600: Ver.01.129 -> Ver.02.104 [Handset firmware] TPA60: Ver.02.02.006 -> Ver.02.04.006 TPA65: Ver.02.02.005 -> Ver.02.04.008 UDT121: Ver.06.04.005 -> Ver.06.06.008 UDT131: Ver.06.04.005 -> Ver.06.06.008 |
Firmware Name: | KX-TGP600 AL |
---|---|
Existing version: | 1.108 |
New version: | 1.177 |
No. | items |
---|---|
1-1 |
Improvement Sending SIP-PUBLISH after each SSRC change during a conversation. Related config parameter: Following settings are needed. VQREPORT_SEND="1" |
1-2 |
Improvement Sending SIP-PUBLISH after each codec change during a conversation. New config parameter: VQREPORT_SEND_OPT_CODEC_ENABLE Related config parameter: Following settings are needed. VQREPORT_SEND="1" |
1-3 |
Improvement Sending SIP-PUBLISH after each network condition change during a conversation. New config parameter: VQREPORT_SEND_OPT_NW_CHANGE Related config parameters: ALERT_REPORT_TRIGGER ALERT_REPORT_MOSQ_CRITICAL ALERT_REPORT_MOSQ_WARNING Following settings are needed. VQREPORT_SEND="1" |
1-4 |
Improvement Improved DTMF impact to MOS score. |
1-5 |
Improvement Shortening the setting time of TR069 SPV |
1-6 |
Improvement Add "Delete handset" function to the web setting. |
1-7 |
Improvement for Dial Plan - dial plan apply for Manual Dial from key pad (post dial, predial) - dial plan does NOT apply for Memory Dial (phonebook, call log, One-touch key dial, Hot key dial) New config parameter: DIALPLAN_MEMORY_DIAL_ENABLE |
1-8 |
Improvement to hide the Call Settings from menu. New config parameter: CALL_SETTINGS_MENU_ENABLE |
1-9 |
Improvement Increase the number of configuration parameters that can be export from the Web setting. |
1-10 |
Improvement Support the Microsoft Windows10/Edge browser. |
1-11 |
Improvement select how to regist the numbers of Outgoing call, which is replaced by Dial plan or inputted. New config parameter: DIALPLAN_REPLACE_LOG_ENABLE |
1-12 |
Improvement : Assign TCP port number randamly in SIP-TLS. |
1-13 |
Improvement : Antenna pictograph and Battery pictograph of TPA60, UDT121 and UDT 131 are changed as follows. Antenna pictograph: It eliminates red frame when the signal strength is very weak. Battery pictograph: It eliminates red frame when the battery level is very low. |
1-14 |
Improvement : Group call pickup allows users to pick up incoming calls within their own group. |
1-15 |
Improvement Support CDP(Cisco Discovery Protocol) |
1-16 |
Improvement Update client-certificate and private-key SHA1 to SHA2. |
17 |
Improvement When a phone checks voice mail, it can send SUBSCRIBE message including mailbox name. New parameter: VM_SUBSCRIBE_SPECIFIC_n |
1-18 |
Improvement A phone doesn't care case-sensitive search of XSI phonebook. |
1-19 |
Improvement Certificate overview is displayable via TR-069 management using Vendor Specific parameter. |
1-20 |
Improvement Add some configuration parameters to separate the update timing of the XSI phonbook data from provisioning. New parameter: XSI_PHONEBOOK_CYCLIC_INTVL XSI_PHONEBOOK_RESYNC_DURATION |
21 |
Improvement When TGP600/HDV begins N-way conference call, the hold music plays for the held outside caller for the entire hold duration New config parameter: CONF_SERVER_HOLD_ENABLE="Y" ( default value is "N". ) |
22 |
Improvement Remote deregistration using new config parameter. New config parameter: NEXT_REGISTERED_HS_NUMBER |
1-23 |
Improvement Support fail over(Redundancy) for Telstra New parameter: - ENH_FOVR_ENABLE_n - ENH_FOVR_RANDOM_TIMER_n - ENH_FOVR_RANDOM_MAX_TIME_n - ENH_FOVR_RANDOM_MIN_TIME_n Related Parameter: - SIP_FOVR_NORSP_n - SIP_FOVR_MODE_n - SIP_FOVR_DURATION_n - SIP_FOVR_MAX_n |
1-24 |
Improvement Display both phonenumber and name even the length overs 16 characters. |
1-25 |
Improvement Changed the rule to display Caller ID which is prioritized to display network information or phonebook information. |
1-26 |
Improvemant TIMEZONE can be set with handset menu. |
1-27 |
Inprovement 1. Device doesn't reboot when resetting line with TR-069. 2. Line status of TR-069 is supported. |
No. | Fixed issue |
---|---|
2-1 |
Modification Fixed issue: If there is not "renegotiation_info" in the sreverHello from the server, TGP600 stop the SSL connection. |
2-2 |
Modification Fixed issue: Failover -Load sharing test case 3 Expected result is that Check that registration is persisted with the same SBC destination. The device must not change to the other equal priority SBC. |
2-3 |
Modification Fixed issue: Delay of audio connection (2 seconds) when answering hunt group call (8 lines x 8 handsets) New config parameter: SIP_INC_INVITE_RTP_MODE_n |
2-4 |
Modification Fixed issue: Server announcement is cut off when user dials *XX for CF feature. |
2-5 |
Modification Fixed issue: There are cases, it lose registration at the constant period when SIP TLS setting is enabled. |
2-6 |
Modification Fixed issue: The device reboots before sending response for setting parameters with reboot process like parameter for VLAN. |
2-7 |
Modification Fixed issue: TR069 process doesn't start when setting ACS server address. |
2-8 |
Modification Fixed issue: There are cases, it lose registration automatically when fail over occurs. |
2-9 |
Modification Fixed issue: Call drops in five minutes when the call is started with CPAT. |
2-10 |
Modification Fixed issue: Browsers are automatically caching the login credentials irrespective of any "remember this password' settings. -> we added Logout button on WEB-GUI |
2-11 |
Modification Fixed issue: Server announcement is cut off when calling to broadworks IVR |
2-12 |
Modification Fixed issue: Incoming call does not stop immediately even though the call is canceled in case of multi line setting. |
2-13 |
Modification Fixed issue: When a phone receives incoming call while calling to the other phone, a user can't cancel the outgoing call on the phone. |
2-14 |
Modification: Fixed issue : TPA65 reboot after change softkey setting. |
2-15 |
Modification When a phone calls and receives '200 OK', call information in PAI header of '200 OK' is not displayed correctly. |
2-16 |
Modification Fixed issue: Because the base unit does not read PIN correctly, when changing PIN of Base unit, there is a case that can not register handset. |
2-17 |
Modification Fixed issue: When you attempt call hold you can hear noise with setting SRTP only. |
2-18 |
Modification We modified to enable to control regarding back light setting which are under below. - Screen Timout - Active level - Inactive level |
2-19 |
Modification Fixed issue: BLF Label name is not displayed correctly in DISPLAY_CALL_KEY_ENABLE="Y". |
2-20 |
Modification Fixed issue: When setting HTTPS_PORT_ENABLE as Y, you can not access to the embedded Web. |
2-21 |
Modification Fixed issue: UDT121/131 reboots when trying to edit the Function key after assigning none. |
2-22 |
Modification Fixed issue: After on hook while line status showing and incoming call receiving, a phone can not goes to ringing status. |
2-23 |
Modification Fixed issue: When displaying RootCA information in Web GUI, display area is too narrow to display all the information. |
2-24 |
Modification Fixed issue: In TPA60, TPA65, UDT1xx, "Line Status" is not displayed. |
2-25 |
Modification Fixed issue: When connecting handset to base unit that does not support CDP, handset reboots by the operation of displaying LineStatus. |
2-26 |
Modification Fixed issue: When the phone receives many incoming calls in a very short period, there is a case the phone reboots. |
2-27 |
Modification Fixed issue: In narrow band setting and handset talk, a phone is slow to be talkable so that head of voice is cut off. (X/M/C only) |
2-28 |
Modification Fixed issue: We modified WEB gui that are effected by CDP. |
2-29 |
Modification Fixed issue: We modified WEB gui that are effected by GroupPickup. |
2-30 |
Modification CSTA bugs are fixed. Fixed issue : - Can't answer incomming call from over 10 digits telephone number. - Call information (Caller / Called) becomes reversed when the call is established. |
2-31 |
Modification Fixed issue : There are ceses where HTTPS provisioning (Client Certificate) fails when the provisiong server is MS IIS. |
2-32 |
Modification Fixed issue : There is a case that the phone does not restart when the restart target parameters has changed in the config download. |
2-33 |
Modification Fixed issue : There are cases where SIP process is deadlocked by sending SIP message and receiving the hole punching packet at the same time. |
2-34 |
Modification Fixed issue : A phone can't answer an incoming call which has phone number over 10 digits on uaCSTA. |
2-35 |
Modification Fixed issue : A phone exchanges information about calling party and called party on uaCSTA after answering an incoming call. |
2-36 |
Modification Fixed issue : STUN configuration is not valid. |
2-37 |
Modification Fixed issue : A user can't input PAUSE when configuring HOT key number even on PAUSE_INPUT_ENABLE="Y". |
2-38 |
Modification Fixed issue : A call is disconnected when an other line is selected while having a conversation even on AUTO_CALL_HOLD="Y". |
2-39 |
Modification Fixed issue of TR-069: Uptime is incorrect. |
2-40 |
Modification Fixed issue : Ring back tone rings against RINGBACK_TONE_TIMING configuration. |
2-41 |
Modification Fixed issue : Ability to program "INT" under softkey. User presses "INT" on stanby screen then selects HS number to call. |
2-42 |
Modification Fixed issue : Ability to check the Base Unit IP Address using UDT Handset. |
2-43 |
Modification Fixed issue : FW update via WebUI will be restricted when Firmware URL is defined and error will be displayed. |
2-44 |
Modification Fixed issue : There are ceses that the terminal does not send invite when it receives refer. |
2-45 |
Modification Fixed issue : There are ceses where HTTPS provisioning fails when the terminal starts up after factory reset operation. |
2-46 |
Modification Fixed issue : Dial plan bug fixed. |
2-47 |
Modification Fixed issue : When importing a config file via WebUI that has any parameter mistake, an error will be displayed. |
2-48 |
Modification Fixed issue : Ability to factory reset TPA65 / 60 when not connected to Base. |
2-49 |
Modification Fixed issue: Parallel Mode Hold Pickup Issue Held call is disconnected when it pickup the call from TPA65. |
2-50 |
Modification A phone continue to seize line even after a call is terminated. New parameter: SHARED_STOP_LINE_SEIZE |
2-51 |
Modification Fixed issue: 1) Once update firmware to default firmware by accessing to Redirection server, below configuration parameters are specified automatically. - CFG_STANDARD_FILE_PATH="https://provisioning.e-connecting.net/redirect/conf/{mac}.cfg" - FIRM_FILE_PATH="http://fw-provi.e-connecting.net/firm/000/HDVx30-yy.zzz.fw" - FIRM_VERSION="HDVx30-yy.zzz" note) yy.zzz is firmware version of default firmware on Redirection server 2) After restart or Power cycle, access to Redirection server. 3) Compare current firmware version with default firmware version. 4) If default firmware version is higher , the phone update firmware. |
2-52 |
Modification Fixed issue: Some words of Russian menu are mistranslation. |
2-53 |
Modification Fixed issue : There are ceses that the transfer function doesn't work when the terminal receives invite including replace. |
2-54 |
Modification Fixed issue : The one way audio occures when the terminal receives invite including replace. |
2-55 |
Modification Fixed issue : There are ceses registration-loss occurs. |
2-56 |
Modification Fixed issue: Handset reboot issue due to NPDI information makes the PAI information longer than 32 characters. Handset now ignores over 32 characters to avoid reboot. (Base Firmware) |
2-57 |
Modification Fixed issue of TR-069: 1. Config file download is failed (Error 9015). 2. Perodic Inform isn't executed when PERIODIC INTERVAL is set more than 24 hours. |
2-58 |
Modification Fixed issue : When importing a config file via WebUI that has any parameter mistake, an error will be displayed. |
2-59 |
Modification Fixed issue of TR-069: Parametrs associated with SIP don't work porperly. |
2-60 |
Modification Fixed issue : In case of TGP600 use TCP and no massage during 120[s], failed INVITE messages (failed with 408 responses back from the SBC) after 120[s] . ->[Modification]the keep-alive time will be changed in the Panasonic phone dynamically based on the SBC expire time received in the 200 OK register. |
2-61 |
Modification Fixed issue : Cancel key is not available when a user is transferring a call. |
Firmware Name: | KX-TGP600 AL |
---|---|
Existing version: | 1.061 |
New version: | 1.108 |
No. | Fixed issue |
---|---|
1 |
Improvement Just after Failover destination is determined, the device sends SUBSCRIBE to Failover destination. |
2 |
Improvement Support blank search in XSI Phonebook. |
3 |
Improvement Enable / Disable Key tone by configuration file. New parameter : - KEY_PAD_TONE_HSy (for TGP600) - KEY_PAD_TONE (for HDV series) |
4 |
Modification Fixed issue: uaCSTA (Mondago / ESTOS) bugs are fixed. |
5 |
Improvement Enhance supported number of Call-ID digits. (Up to 128 digits) |
6 |
Modification Fixed issue : - One-way call is happened in specific server when answering the call. (MULTI_NUMBER_ENABLE="Y") |
7 |
Modification Fixed issue : - Start call duration count from receiving 183 early media. |
8 |
Improvement MoH(RTP 1) and Hold tone (RTP 2) are not mixed even when server send 2 different RTP to the phone. |
9 |
Improvement Support DNS-SRV Lookup for XMPP (UC-One). New parameter : - UC_DNSSRV_ENA - UC_TCP_SRV_PREFIX |
10 |
Modification Fixed issue: Countermeasure of following DTMF issue. When a TGP600 handset sends DTMF in hands free mode, the handset's mic pics up the tones and transmits them. The receiving party hears each inbound DTMF twice. |
11 |
Modification Fixed issue: Countermeasure of following DTMF issue. During 3-way conference which is established by TPA60/65, DTMF from the handset will send to one conference party only and not the other. |
12 |
Modification Fixed issue: Regarding SIP-PUBLISH. - SIP PUBLISH is not sent when SSAF setting is enabled (SIP_DETECT_SSAF_n="Y"). - Improvement of the header format of SIP PUBLISH. - Calculation method of MOS-CQ value which is contained in the VQReport of SIP-PUBLISH has error. So the calculated value is always too low as compared to actual voice quality. |
13 |
Improvement: MOS-CQ value of SIP PUBLISH which is sent after a disconnected call is calculated from the MOS-CQ value of the last RTCP of a conversation. It is improved to calculate the SIP PUBLISH MOS-CQ value by the average of all RTCP from the entire conversation. |
14 |
Modification Fixed SIP TLS issue. After establishing a call by using SIP TLS, place the call on hold and then resume it, - Operation side (TPA60/65) : Noise burst is heard. - Other side (intended party) : Just silent. |
15 |
Improvement: Through provisioning, a time range can be defined using a new configuration parameter to randomly access the server to get firmware updates. New parameter: - FWDL_RANDOM_DURATION Related Parameter: - FIRM_UPGRADE_ENABLE - FIRM_FILE_PATH - FIRM_VERSION |
16 |
Improvement Handset registration improvement. Registering a new handset to the base station fails intermittently in crowded DECT environments. |
17 |
Modification The device continues to send DNS query, when the DNS server lapses into unusual condition. |
18 |
Modification The hold tone generated inside of the device is mixed with the received announcement, when it is connected to the announcement server of Gneband. |
19 |
Improvement: Through provisioning, a time range can be defined using a new configuration parameter to randomly access the server to get configuration file updates. New Parameter: - CFG_RESYNC_DURATION Related Parameter: - CFG_STANDARD_FILE_PATH - CFG_PRODUCT_FILE_PATH - CFG_MASTER_FILE_PATH - CFG_CYCLIC - CFG_CYCLIC_INTVL - CFG_RESYNC_TIME - CFG_RESYNC_FROM_SIP - CFG_RESYNC_ACTION |
20 |
Modification Fixed issue : Send no voice RTP in specific server. |
21 |
Improvement: The phone number of each registered SIP Line can now be verified in the handset menu [MENU} -> [System Settings] -> [Status] -> [Line Status] -> select line |
22 |
Improvement Call can't be transferred to Conference server when user part of CONFERENCE_SERVER_URI is longer than 63 characters. |
23 |
Modification Fixed issue : - When the phone receives Re-INVITE before sending 200 OK as a reply for the initial INVITE, the phone sends Request Pending and call is disconnected. |
24 |
Modification Fixed issue: Setting change of the port number of Line 1 does not affect immediately under the SIP Trunk condition. |
25 |
Modification Fixed issue: The phone could not resume the paired SCA phone's hold call when it is connected to MetaSwitch. |
26 |
Improvement AMR-WB description in SDP is modified. |
27 |
Modification Fixed issue : Czech letters can't be displayed in XSI Phonebook. |
28 |
Modification Fixed issue : The phone can't download configuration file from option66, when fail download from option160. |
29 |
Improvement The phone can receive MCP(Polycom formula) more than once. |
30 |
Modification Fixed issue : The phone can't receive configuration file from option66 in while pre-provisioning , when there is conflict with DHCP request timing. |
31 |
Modification Fixed issue : Unnecessary characters are displayed in the line name during the operation of the voice message confirmation screen. [ Handset Firmware Version ] -TPA60: Ver.02.02.007 -TPA65: Ver.02.02.006 -UDT121: Ver.06.04.006 -UDT131: Ver.06.04.006 |
32 |
Improvement Add Display Name for LocalID and OrigID for incoming call in VQ Monitor Tool. |
33 |
Improvement Display the version of Panasonic SIP software on WEB. The version is managed in common over the models (HDV series and TGP600). |
34 |
Modification Fixed issue: Phonebook search with LDAPS is failed.(TGP600 FW version01.094 only) |
35 |
Improvement HTTPS protocol can be used for Web GUI for secure reason. New parameter - HTTPS_PORT_ENABLE |
36 |
Improvement for supporting SIP-Trunk -DID(Direct Inward Dialing) function is supported. -SIP server of no registration type is supported. New parameter -SIP_TRUNK_MODE_ENABLE -SIP_NON_REGISTER_ENABLE |
37 |
Improvement for SIP-Trunk service Hold tone can't be sent from TGP600 when MOH server isn't used in SIP Trunk service. Adding "Send RTP on Call Hold" to Web menu. New Parameter: -RTP_KEEP_ENABLE |
38 |
Modification Fixed issue : Subscription-State in NOTIFY sent by the phone is missing. It causes REFER transfer fail in specific server. |
39 |
Modification CSTA bugs are fixed. Fixed issue : - Can't answer incoming call from over 10 digits telephone number. - Call information (Caller / Called) becomes reversed when the call is established. |
40 |
Improvement Update client-certificate and private-key SHA1 to SHA2. |
41 |
Modification Fixed issue : There are cases where HTTPS provisioning (Client Certificate) fails when the provisioning server is MS IIS. |
42 |
Modification Fixed issue : There is a case that the phone does not restart when the restart target parameters has changed in the config download. |
43 |
Modification Fixed issue : There are cases where SIP process is deadlocked by sending SIP message and receiving the hole punching packet at the same time. |
44 |
Modification Fixed issue : A phone can't answer an incoming call which has phone number over 10 digits on uaCSTA. |
45 |
Modification Fixed issue : A phone exchanges information about calling party and called party on uaCSTA after answering an incoming call. |
46 |
Modification Fixed issue : A phone can't communicate with a cell phone even when AMR-WB is configured. |
47 |
Modification Fixed issue of TR-069: Uptime is incorrect. |
48 |
Modification Fixed issue : There are cases where HTTPS provisioning fails when the terminal starts up after factory reset operation. |
49 |
Modification Fixed issue : There are cases registration-loss occurs. |
50 |
Modification Fixed issue: Remote ID information of the first call for SIP Publish is changed to remote ID information of the second call when completing ghee transfer of originating call. |
Firmware Name: | KX-TGP600 |
---|---|
Existing version: | V01.027 |
New version: | V01.061 |
No. | Fixed issue | |
---|---|---|
1 |
Improvement to add PTIME 60msec to RTP ability. |
|
2 |
Improvement to output the DTMF in both the SIP-INFO and Outband. |
|
3 |
Improvement to hide the anonymous menu. |
|
4 |
Modification - Time setting does not work with POSIX time zone. |
|
5 |
Modification - Device does not send subscribe for provisioning. |
|
6 |
Modification - User can not call if device has received two kinds of RTP under RTCP-XR function ON. |
|
7 |
Modification - WEB does not open, when device is set to HTTP_PORTAUTO_OPEN="Y" for a long time. |
|
8 |
Improvement to prepare the option to prevent web setting. |
|
9 |
Improvement - The device allows "&" character to the URL of the XMLphonebook. |
|
10 |
Modification - Corresponding to the config of tone gain. |
|
11 |
Modification - Server disconnects SIP-TLS connection in five minutes between the Device and the Server. |
|
12 |
Improvement - Changing the web word "Display Name(LDAP_BASEDN)". |
|
13 |
When you release the hold with RTCP under the control IP-PBX, device become to silence or reboot. |
|
14 |
Improve the SIP register stop after received the bubble UDP packet. |
Firmware Name: | KX-TGP600 |
---|---|
Existing version: | V01.008 |
New version: | V01.027 |
No. | Fixed issue | |
---|---|---|
1 |
The INVITE message contains incorrect FROM Header, when PHONE_NUMBER_n and SIP_URI_n (e.g. 41319180076@swisscom.ch) are set. The FROM Header contains domain name doubly as follows. - issue From: <sip:+41319180076@swisscom.ch@swisscom.ch;user=phone>;tag=53169100 - after the countermeasure From: <sip:+41319180076@swisscom.ch;user=phone>;tag=53169100 |
|
2 |
Infrequently FWD/DND does not cooperate with the settings of WEB of Broadsoft. |
|
3 |
The device does not display the name of intended party, when it is received incoming a call that contains valid PAI information. |
|
4 |
XSI Remote Office and XSI Any Where do not work, in case of using HTTPS. |
|
5 |
Searching of XML phonebook does not work. |
|
6 |
It takes about 3 seconds to update status of WEB. (countermeasure for the issue pointed by PSCNA) |
|
7 |
Infrequently the device does not start sending RTP packets when it is received 183 message instead of 180. |
|
8 |
Regarding the function of uACSTA, the consultation transfer does not work correctly. |
|
9 |
Improvement of the voicemail function. Permit SIP URI as voicemail number (VM_NUMBER_n). |
|
10 |
Improvement of valid digits of Display Name. Even if it is set 32 digits of display name via WEB of Broadest, the device becomes to display within 24 digits as its Display Name (DISPLAY_NAME_n). |
|
11 |
Improvement of the specification of pre-provisioning. The change of the implication regarding Option 66 and 67 of DHCP. |
|
12 |
Improvement of supporting RFC3264. The device becomes to be held a call from other SIP phone supporting RFC3264. (request from BS Cloud service of UK ) |