GXP-2000 Legacy Firmware Notes

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Due to the main GXP-2000 page becoming overcrowded with notes about old beta and other legacy firmware versions, that information has been moved to this page. - thetatag








Firmware Notes 1.1.5.15:

Release Notes:

Build 1.1.5.15 (11/19/2007)
• Fixed GXP display "Not Registered" for accounts that does not require registration
• Fixed GXP1200 LCD display corrupted
• Fixed AEC divergence during call waiting tone
• Fixed default router gets changed under static IP mode incorrectly
• Fixed GXP1200/GXP2010 out-of-memory problem/idle screen crash. Note: A hard power-cycle is REQUIRED after the new firmware is loaded.
• Fixed softkey problem with onhook dial
• Fixed softkey problem with direct IP call
• Fixed GXP2010 color LED blinking issue
• Fixed GXP2000 EXT issue
• Fixed BT200 redial problem after deleting called log. Fixed GXP missed call cannot be removed by LEFT key completely
• Added support for padding bit in RTP
• Fixed GXP2010/GXP2020 LCD backlight does not function correctly
• Fixed BT200 change codec from UI MENU will also change Voice Frames Per TX incorrectly


Post your experiences here:
  • Major : Phone crashes whilst on mains power, leading to situation where phone will only boot without a patch cable plugged in, has happened to 30% of our handsets within 18 months. Changing power and/or network cables does not fix the issue.- SummerIT
  • Major : Here's a new one for you to puzzle over... the phone is registered OK, shows the IP address, clock is correct, clock cursors flash, but when picking up handset, or pressing "speaker" there is no dial tone - but the speaker symbol appears at the top of the screen, I can enter numbers, they appear in the in the display, I can hear tones when pressing numbers, but pressing "send" or waiting for it to time out has no effect - the speaker symbol shows, but it doesn't dial. HV1.1A, FW1.1.5.15. PITA2.0 - olly
    • Confirmed. I'm also having this issue... EXTREMELY annoying, because people can't reach me, although for me everything seems all right. - mirak
    • Note : I have had this same issue with a GXP2010 running FW1.1.5.15. The phone still shows as registered to the PBX and receives MWI updates. It will not receive or make calls. When trying to place a call, the LCD screen shows some incorrect formatting, such as words written on top of one another. Seems to happen once the phone has been up for about three to four days. - rconner1890
  • Major : Why oh why do I try? Busy tone at exactly 2 minutes into the conversation, then disconnects at exactly 2.5 minutes. Nothing seems to make this piece o crap happy! :( - indesignfirm
  • Major : My phone (GXP2000) just rebooted in the middle of a conversation. - doesn't seem to be deterministic, but it's quite annoying, as my phone's in production use - and not test use :( - is the previous 'stable' version still available somewhere, so I may try and downgrade? Random reboots of the telephone are just unacceptable... :( - kodomo
    • Note : Me too! two out of five GXP2000 with this software restart at random, approx every 24 hours - olly
    • Note : Kodomo and Olly, please state the Hardware Version of your phones. I have seen that older HW versions seem to have a lot of issues with modern firmwares. Check hardware verison by entering phone menu, selecting "status", then scrolling all the way to the end and looking for a string that starts with "HV:" - Kurgan
      • Note : GXP 2000 HV1.1A - olly
      • Note : To me, it seems completely random (not even the 24h) I haven't been able to reproduce it yet - HV:v0.4 - kodomo
        • Note: Kodomo/Olly, please contact Grandstream for this, this issue should be addressed. Before that, you can try "gray"s magic, disable watchdog and see if it works
          • Question: How (may) do I disable watchdog permanently? - GMaoret
          • Note: After their spontaneous rebooting in days 1,2 and 3, my phones have not rebooted now for between 6 and 10 days - could it be that they download extra bits of firmware in the first few days, which requires a reboot? - olly
          • Note: For me it's the same, although I don't believe it's a download, since I disabled downloads again immediately after my update. I'm quite puzzled by this, but I've been unable to reproduce the reboot issue (even trying 2-3day long calls - incoming and outgoing)... I guess it's a rare race condition in my case(?) - kodomo
      • Note : HV:1.1A No problems with the call disconnection - nonsequitir
    • Note : Me too, reboots spontaneously on or off call ~once per day. disabling watchdog under factory settings seems to have an effect. HV:1.1A -p
  • Major : Two identical GXP-2000 + EXT stopped showing extension status on the EXT module. Transfers originating from the EXT module buttons where also triggering another extension to ring in addition to the desired extension. Don't know what triggered that condition but a reboot of the phones settled the problem. Both phones were up for 8+ days (since install) before this happened. Will wait another 8 days to see if this returns. HV is 1.2b - jflaplante
  • Major : GXP2000 (Hardware 1.1A) Call self-terminates after about 30 seconds when on HANDS FREE with MUTE enabled (Ticket#: 2007121199000214) - nonsequitir
    • Note: Grandstream support have recommended that I set "NAT Traversal (STUN)" as "No, but send keep-alive" - nonsequitir
    • Note: gray, you may want to check if the call gets established, from console, if still not working, check with Grandstream Support.
    • Note: Me too! This has been driving me nuts for as long as I've had this phone. I do a lot of conference calls, and inability to mute on speakerphone sucks. HV:0.4 - mjmac
  • Minor While this is not a new issue, it should be addressed. When logging into the web interface with the end user account, the Ext modules cannot be accessed. I believe setting button number options is an end user feature, not one for an admin. - bmullins
  • Feature Request : Add the ability to adjust microphone volume level. I thought this use to be a feature, maybe i got it confused with my snom 360 which allows this basic feature. Some gxp-2000 are way to loud to the other caller. - mike240se
  • Major : Language Pack 1.3 is unusable. Language file is named gxp2000.lpf instead, and if renamed to gxp.lpf the phone cannot write it. Phone's syslog shows: "ERROR write gxp.lpf" - Alex
  • Major : I have two GXP200s which have been rebooting quite often since I installed 1.1.5.15, mostly seemingly randomly during calls, but recently one started rebooting every time I woke my adjacent MacBook Pro from sleep, which I think I narrowed down to happening when the Airport wifi adapter was enabled and not otherwise. I've just disabled WDT and if that doesn't fix the rebooting I'm getting ready to throw the phones out. Any recommended replacements? adw 2008-02-03 Disabling WDT may have been a red herring. The ethernet out cable for connecting PCs seems to be a major contributor to the random reboots. Without that cable, the phones are less susceptible to random reboots, but still not completely reliable. And it's a pain not to be able to use that port, so still looking for a replacement adw added 2008-02-21
  • Major: After upgrading to 1.1.5.15 I have a few different issues. We are experiencing call dropouts after a couple of minutes (crackle/screech then dropout) MinorOn the interface, I cannot change optional rule in ntp, it always comes back to the default - annoying as it means I cannot make it display the correct time for South Australia - even though it is configured by both DHCP and NTP. Also I cannot get it to always use the lcd backlight. again, it just wont change the setting via the interface tmcruisin 04-03-2008






Firmware Notes 1.1.5.10 (Beta):

Release Notes:

Build 1.1.5.10 (11/01/2007)
• Fixed audio stack overflow crash (A1000 crash problem found on GXP1200)
• Fixed GXP2010/2020 LCD contrast MENU OK/CANCEL soft key
• Fixed onhook-dialing account selection bug
• Fixed GXP1200/2010 TFTP display incorrect model name and remove hyphen in all model names
• Adjusted MWI flash interval to 1 second between on/off
• Removed eventlist URI settings from account pages as GXP1200 does not have BLF keys
• Added "LCD Backlight Always On" and "Disable Missed Call Backlight" to GXP1200 Basic Settings page
• Changed GXP20xx speed dial key modes and GXP2000 EXT key modes from "Asterisk BLF" to "Busy Lamp Field (BLF)"
• Fixed BT200 redial problem after deleting called log
• Fixed GXP1200 does not reject 3rd incoming call correctly leading to display problems
• Fixed GXP1200 web page problem after 3 concurrent calls
• Fixed GXP1200 busy call forward not working
• Fixed GXP1200/2010/2020 setting DHCP mode in GUI MENU when already in DHCP mode causes it to toggle to static IP mode
• Fixed GXP2010/GXP2020 provisioning screen display not aligned/missing in 1.1.5.6
• Implemented HEADSET key logic so it works similar to SPEAKER key
• Improved handset duplex performance


Post your experiences here:

  • Major : Provisioning using a full template file (as supplied by Grandstream) sends the phone into a constant reboot cycle until the provisioning file is removed, removing P342 (Display Language) as reported below (and a number of similar values) does not fix. The provisioning file does not crash earlier firmware versions such as 1.1.2.25 - Grandstream Ticket 2008020799000203. - Rob
    • Grandstream support indicated that this issue has been fixed in firmware versions 1.1.6.7+. As a temporary workaround they suggested removing the 'P' values for multipurpose keys 8-18 in the template, which worked for me. They further recommended only specifying the 'P' values you wish to change from factory defaults in the template file. - charkins
  • Major : BLF does not work with Asterisk 1.07 from Debian Sarge. Every version past 1.1.1.14 does not work. This is annoying, since I cannot upgrade every Asterisk server I manage. - Kurgan
  • Major : BLF is still not working with asterisk 1.4. In other words BLF is not working on asterisk 1.4 for all GXP-2000 firmware. That's really, really annoying. - Dusty
    • Note : No BLF problems here with Asterisk 1.4 with GXP2000 and GXP2020 with this version. I think you have a problem with asterisk (are call-limit and limitonpeers set in your sip.conf?) - francesco_r
      • For francesco_r: Yes I tried those options like 100 other combinations, no way to show a peer unreachable/unregistered. Please contact me at dustyATnibblesDOTit if you can help, I'll share experience if solved - Dusty
        • This is a bug in asterisk, not grandstream: i have opened an issue one month ago in the the asterisk bug tracker: http://bugs.digium.com/view.php?id=11025 - francesco_r
    • Note : BLF has always been working with Asterisk 1.4. You're probably not setting call-limit and limitonpeers. See: http://www.grandstream.com/documents/GXP2000BLFwithAsteriskConfiguration.pdf, bottom of page 2.
  • Major : no dial tone after sip server not responding for long time, if i call the phone asterisk say "internal server error", i must reboot the phone (the changelog of release 1.1.5.6 report that was resolved but i still have this problem with both gxp2000 and gxp2020) - francesco_r
  • Major : Still unabled to change the GXP2020 Gateway IP if the phone is set with a static ip address. This was supposedly fixed in 1.1.5.6. The address still revert back to 192.168.255.0 even if I set the IP in the config file - Matt Gorecki
  • Major : Phone keeps rebooting after getting provision file if option P342 (Display language) is set. This problem is seen on all version from at least 1.1.4.x. - Bad2Dbone
  • Minor : The italian language pack is wrong, it's german! - francesco_r
  • Minor : Be nice if TRNF key could be edited. - ymartin
  • Minor : Just my gxp2000 was rebooting at least twice a day on this firmware seemingly at random. Never had this problem before and all of our other phones are on the same firmware with no problems. Tried factory reset but did not help. I solved the problem by disabling the watchdog timer. My phone has now been up for 4 days with no problems. I spoke too soon, after 4 days wigged out (all lights strobing very quickly) during a call (ulaw).. Went back to 1.1.4.22 -p
  • Minor : on GXP-2020 the Dialed calls display 24 times the last call !!- furio
    • Note: I cannot reproduce this issue, my 2020 shows dialed calls correctly - Kurgan


Firmware Notes 1.1.5.6 (Unreleased Beta):

Release Notes:

Build 1.1.5.6 (10/22/2007)
• Slow key response fix
• Added GXP1200 LCD Contrast Control
• After Onhook dial the # key and default 4 seconds send number feature cannot work
• keep the PPPOE setting cause the static ip mode setting fail
• Periodic NTP Re-sync not working properly
• 3-way-conference under Notrl MCS mode is not working
• Added syslog if we are unable to allocate anymore channels
• Fix if the server in DNS form is configured, the provision will start two times
• Added "LCD contrast" into the language string
• Updated the new bootloader application status messages to show tftp download and writing messages.
• Implemented vertical status line for gxp1200 for three line display when offhook + softkey.
• For gxp2020 and gxp2010 - cannot change Gateway IP address in static ip mode
• Ported RFC call pickup to be enabled in all GXP
• Changed "Asterisk BLF" string to "BLF"
• Added new menu to reset VLAN under "config"
• GXP2020 offhook screen clears the whole main display area
• Fixed phonebook does not Re-Download atthe specified interval
• BT200: Mute ringer in headset does not work in 1.1.4.20
• Fixed turn off speaker remote disconnect feature for BT200.
• Added two menus to set VLAN tag and priority from the phone
• Fixed: Check SIP User ID for incoming INVITE" checks To header instead of Request URI
• Devices behave incorrectly while in call on receiving NOTIFY (Event: check sync)
• Fixed an issue when the call is on hold, the SOFTKEY is still in OFFHOOK mode.
• Added LCD contrast menu under preference for gxp2020 and gxp2010.
• LCD contrast setting will cause no change if it is not exited properly.
• Added ok and cancel SOFTKEY support for LCD contrast menu.
• GXP2000 1.1.5.3, no dial tone or LED on line key, after the SIP server not responding.
• Presence/BLF button also function as SPEEDDIAL button when idle
• 200 OK for video 'm' line contains an extra space character at the end of the line
• Fixed gxp1200 LED flicker issue



Build 1.1.5.3 (Beta):

Release Notes:

Build 1.1.5.3 (09/20/2007)
• Audio issue on GXP2020
• Added on-hook dialing
• Decoupled registration and subscription intervals
• Fixed we do not re-transmit 200 OK for reINVITE
• Fixed call history time not adjusting to DST
• Support dialing from phonebook with SEND key
• Support for disable backlight when there is missed calls
• Support for BLF call pickup using dialog automatically if XML contains sufficient information
• Support XML application for GXP2020
• Enhanced GUI including reversed MENU, larger fonts for GXP2020
• Integrate my new HLC code + handset gain tricks to reduce ambient noise picked up by handset and eliminate clipping of loud audio
• Remove extraneous lock around packet dequeue loop
• To stop network flood from preventing call setup (tx path works fine in this condition, rx path stutters due to current ne2000 receive buffer overflow handling).


Post your experiences here:

  • Added page up/page down buttons in menus
  • Increased size of all screen elements (less whitespace)
  • New option in 'Basic Settings': 'Disable Missed Call Backlight:'
  • Phonebook Alpha Scroll: Push the corresponding number on the keypad and it will jump to that letter in the list. i.e. press 6 twice to get to the N's in the phonebook! - Shane Steinbeck
  • Audio Quality: I'd say the handset sounds about as good as it has in the last year or so. I think the speakerphone still sucks...I've pretty much given up on that ever working. I bought Aastra 57i's and the speakerphone is flawless. Did I mention I bricked a GXP-2000 and lost more in a flood? Only 4 GXP's remaining :-) - Shane Steinbeck
  • NOTE: Audio quality is improved, I can hear everyone clear with using G729 codec (mnam)
  • On-hook dialing - bcheath

Major : (10/11/2007) On one phone the display is all wrong, the main page is a bit wrong, but the other pages are unreadable. - Richard
    • Note : I can confirm the display problems on my GXP-2000 with build 1.1.5.3, HV: 0.3. I was able to downgrade to version 1.1.4.25 - Knut-Helge
    • Note : I could not confirm any display problems.National characters shows correctly, I have HV:0.4 - Mnam

Major : (10/13/2007) Phone hang ups sometimes without any reason. (ping accessible,but no web + no response from display) - mnam

Major : (10/23/2007) Phone was locked up when I arrived in the office this morning. rebooted. Phone started booting on its own several times later on. It took several minutes to register. Finally it would not register at all. I reinstalled FW 1.1.4.18 - it seems back to normal. (HW=0.4) - - notoncloudnine

Major : (10/24/2007) DTMF RFC2833 stop to works after a transfer. I must set asterisk to INBAND to make DTMF work reliable. The other previous versions worked well. (HW=1.2B) - - francesco_r

Major : (10/24/2007) BLF still unreliable. Sometimes the led is blinking or red when an extension is not in use, and asterisk show correctly the hint idle (HW=1.2B) - - francesco_r

Minor : (10/25/2007) "No Key Entry Timeout" stop to works after the first Onhook dialing (HW=1.2B) - - francesco_r


Firmware Notes 1.1.4.22 (Beta):

Release Notes:

Build 1.1.4.22 (08/31/2007) ----

  • Fixed RTP fails when VLAN is used
  • Handset improvement + packet handler optimization
  • Add Syslog message for off-hook/on-hook notification
  • Fixed we incorrectly drop URI parameters in Refer-To header
  • Fixed GXP2020 one-way audio problem
  • Fixed we incorrectly turns on media when received 18x from caller
  • Fixed we incorrectly turns on media when received INVITE with Replaces and the call its replacing is still in RINGING state
  • Fixed when configured to use STUN and no server is configured we would be re-registering every 30 seconds
  • Added support for early attended transfer (A talk to B, A put B on hold and call C, while C is running A transfer to B) ----
  • Added a "*" suffix to User-Agent field if provider lock is enabled
  • Added audio soft-clipping in TX path
  • Fixed GXP2020 MENU problem reported (Upgrade menu)
  • Fixed BT200 crash on off-hook reported
  • Fixed GXP SIP passwords not stored correctly on additional accounts
  • CN packet handler and generation
  • Fixed random crash problem found in 1.1.4.x ----
  • Changed all GXP to use gxp.lpf to share multi language file
  • Fixed custom idle screen some BMP displayed reversed color
  • Fixed GXP display incorrect EXT key event in factory diagnostic mode
  • Fixed GXP cannot use speed dial key as line keys when all line keys are in use
  • Fixed we turn off speaker on remote disconnect regardless of the setting
  • Added option to change SIP Transport in MENU
  • Added option to configure SIP settings for additional accounts on GXP
  • Added validation check for subnet mask, we will ignore invalid subnet mask
  • Fixed iLBC, G.722, and G.729 problems
  • Fixed out-of-order packets when calling Asterisk voicemail
  • Eliminated the idle screen crash and occasional boot up crash I was still seeing (on 1.1.4.17/1.1.4.18)
  • Fixed AGC does not update TX digital gain, enabling speaker phone to work at longer ranges

  • Louder maximum volume for AER
  • Fixed when a value of 256 or higher is specified in one of the octets for subnet mask, it will be treated as 255. If anyone octet is invalid, the entire address is ignored. This is applicable to Preferred DNS server, IP address, default gateway, DNS Server 1, and DNS Server 2 fields.
  • Changed if invalid (outside 96-127) iLBC payload type is specified we ignore the invalid value (we use to set it to 98)
  • Support for provider lock option
  • Added BT200 display calling number for offhook auto dial
  • Changed Upgrade MENU so changes are saved immediately when OK is pressed instead of needing to choose another Save option
  • Fixed headset volume issue


  • Major : (Oct30/09) BLF is not working with asterisk 1.0.7 from Debian Sarge (old stable). - Kurgan
  • Major : (Sept24/09) BLF is not working with asterisk 1.4.7+ - Dusty
  • Minor : (Jul25/07) DND and BLF broken If DND is activated on a phone using the MUTE/DEL button and the phone is phoned from any another extension also GXP2000, the BLF corresponding to the extension dialed keeps on flashing RED on all phones including the one dialed even after going on-hook. - theobresler
  • Major : (Sep21/07) __Random Lockup. We've had one lockup in 2 weeks and keep the phone sending data to syslog. No message in the log and had to cold boot the phone. Better than previous firmwares, still not perfect. - InDesign Firm
    • Note : (Sept21/07) I can confirm the Random Lockup. I am seeing intermittent (twice in as many weeks) lockups on one GXP2000 that had been perfectly reliable running 1.1.1.4. Oddly, only one of our three phones ever "froze" (requiring a cold reboot / power cycle to restore operation) with 1.1.4.18. The same phone is now locking up under 1.1.4.22 (I never installed 1.1.4.20). The phone that's locking up is the newest hardware of the three and is very seldom used. For this reason, it seems as though this lockup symptom may somehow be related to hardware or the usage (or lack thereof) pattern.- engineerdan
  • Minor : (Sep/07) G.722 problems GXP-2000 using G.722 still has audio glitches. Transmitted audio (from phone) drops out or mutes. Tested with Asterisk and Polycom G.722 - andrew
  • Minor : (Oct/07) BLF does not function when Send Anonymous set to YES under Account - justind





Firmware Notes 1.1.4.20:


This firmware can be downloaded from http://www.bobbyspad.com/

Bug Fixes:



  • Minor : (Jul27/07) Custom Screen Reversed We have built a logo and custom screen for one of our customers. Loaded the firmware and now the screen is reversed.. - InDesignFirm



Firmware Notes 1.1.4.18:

This firmware can be downloaded from the official firmware page http://www.grandstream.com/firmware.html

Build 1.1.4.18 (07/17/2007)
  • Note: Firmware known to lockup on boot if SIP set to use TCP instead of UDP - see notes below

Bug Fixes:

  • Fixed BT200 does not return to idle screen upon attended transfer complete
  • Fixed GXP line number display bug when line number >= 10
  • Fixed GXP2020 WEB UI displays incorrect default SIP port number for Account 5/6
  • Fixed choppy GSM audio quality
  • Fixed G729/G723 may crash in-call

  • Major : (Oct30/07) BLF is not working with asterisk 1.0.7 from Debian Sarge (old stable). - Kurgan
  • Major : (Sept24/09) BLF is not working with asterisk 1.4.7+ - Dusty
  • Major : (AUG17/07) Firmware Release Problems So in 2 weeks worth of conversations with Grandstream Technical support, I finally had them tell me that 4.18 has many known problems and that they have sent it back to their development team for changes. According to the rep I spoke with on the phone, they do not have any new available beta firmware, as they are trying to fix tooo many big problems... I was also provided with a newer version of the firmware for my GXW4104, but was warned that if I installed it, and it didn't work, I couldn't downgrade. Good to see I'm being presented with so many options. On a side note, purchased an Aastra 57i and a Cisco phone. Their is no comparison on how much better their sound quality is. They are absolutely the shizzle. I guess you get what you pay for. - InDesign Firm
  • Major : (Jul31/07) Lockups & Bad Sound Quality We've upgraded several of our GXP2000's only to find that we had a horrible echo when using it with the GXW4104 gateway. We've rolled back to 17 still have it, so rolled back to 4.14 and it has gone away. We've also noticed that in 18 if you set the packets from UDP to TCP the phone just locks up on the next reboot and won't let you get in at all... Had to disconnect from network, reboot, and try to type in the MAC address before it locked up again to reset to factory, just to be able to reload new firmware... I think I'm gonna try an Aastra phone! This is frustrating... - InDesign Firm
    • Note :(Aug20/07) SIP TCP Lockups Configuring to use SIP over TCP will cause the phone to lockup/hang while trying to connect over SIP. DO NOT change SIP over TCP. InDesign's instructions for fixing a locked phone follows. - Tim Smith
      • Note :(Aug23/07) SIP TCP Lockups Unlocked, KINDA The way we were able to recover the phone was to unplug the network cable. Reboot the phone, and then use the on screen menu's to reset to factory settings. At that point, plug back into network, get address, program. If we left it plugged into the network, the second it tried to register, the phone would just freeze. We were never able to get it to work with TCP. You really shouldn't use with TCP unless you are having routing issues, but we were just trying all buttons and options when we found it killed the phone. - InDesign
    • Note :(Aug2/07) Lockups We have 5 GXP2000's, upgraded over the past week to firmware 18, and experience random lockups when idle. Some of them locked after installing the upgrade (from 14), all of them have locked at different times of day - sometimes during office hours, sometimes middle of the night, but only ever one at a time. When they lock the screen appears as normal, the only clue to the lockup is the clock showing the wrong time, and, of course, unresponsive to any keys or incoming calls. Solution is to pull the power and reboot. - olly
    • Note :(Aug5/07) Bad Sound Quality We've upgraded 3 phones to 4.18 and have the same echo problems with GXW4104. I tried all different echo and TX silence settings to no avail. I'm sick of battling and rolled back to 4.14 as well. I must admit that, for future phones, I am also definitely removing Grandstream from my list of possible phones or FXO gateways. - theobresler
  • Minor : (Jul27/07) Ringtones distorted I've made custom ringtones that are bits and pieces of songs. With this firmware, they sound distorted. They sound much better in previous firmwares. (is this because the speakerphone echo canceller?) (HW=0.4).- Joe M
  • Minor : (Jul26/07) Ringtone volume is much louder now. Even at the minimum level it is way too loud when you're sitting just a feet away from it. It's disturbing to others in the same office too. - forsen
  • Minor : (Jul25/07) If DND is activated on a phone using the MUTE/DEL button and the phone is phoned from another extension also GXP2000, the BLF corresponding to the extension dialed keeps on flashing even after going on-hook. - theobresler
  • Major : (Jul24/07) Phone disconnects after 5 minutes after Mute has been actived (HW 0.4). Use case: Listening to a presentation on speakerphone through a conferencing provider. Not sure if the provider disconnects because of 5 minutes of absolute silence from my side? - kam - Disable VAD on your phone. Mute+VAD=No RTP. Most VoIP servers will disconnect because of lack of RTP. - andrew
  • Major : (Jul23/07) Phone freezes in call Using PCMU, HW 0.4 - ninthclowd
    • Note : (July24/07) I have not seen this problem at all on 30 phones. How do you reproduce it? (HW=0.4).- anthony
      • Note : (July26/07) Anthony, as I said, it "happens in call" which indicates that it "randomly happens in call without any sort of catalyst". If you are a GS tech then give me a call (Bill Zhang has my number) and I will get you my config files and screen bg image - ninthclowd
    • Note : I can confirm that the GXP2000 does freeze while in a call using G729, HW 0.2 - bjefferys
  • Major : (Jul26/07) Phone freezes while idle Using PCMU, HW 0.4. I never had this happen before in 1.1.14.17 but it just happened today- ninthclowd
  • Major : (Jul20/07) Breaks BLF light on GXP2000 BLF light is blinking after a call for up to 30 minutes.
    • I can confirm this, i had the problem with both GXP2000 and GXP2020 - francesco_r
    • I have the same problem with the GXP2000 and Asterisk BLF. Other than this the phones rock. -p
  • Major : (Jul23/07) Speakerphone still useless Using PCMU, HW 0.4. various buzzing and volume changes during call - ninthclowd
    • Note : (July24/07) My experience is that speakerphone is vastly improved over 1.1.3.X series, , and I am quite happy with it. You must make sure that you dont mix f/w versions among your phones, as this screws up speakerphone. (HW=0.4).- anthony
      • Note : (July26/07) Anthony, phones should connect to a server and then to another phone in most environments, not by direct IP. This being the case, feedback,buzzing, hissing, etc. are present when doing an echo test to the asterisk server, but only when on speakerphone. This indicates a problem with the phone or the server. In my installation I have many different makes and models dialing outbound through my server with none of these feed back problems. This indicates that I do not have a problem with my server, which makes sense since I am running the latest stable distro of asterisk/zaptel/etc and have tuned my gains to the local telco test line. So the problem is with this particular phone sitting on my desk. Unlike you, I have no desire to move my entire user-base to a beta firmware(that you can't downgrade from) just to fix my speakerphone issues. In the future I would appreciate you keeping your opinion to yourself instead of policing my opinion of whether or not there is something wrong with my phone, newb - ninthclowd
        • Note : (August12/07) After upgrading to .18 from very old firmware that came preinstalled on phone, the speakerphone is much improved. Previously, callee could not hear, while caller could. Now, I have had several successful calls using the speakerphone. And, touch wood, no other problems. However, my setup is very simple, just direct connection to four VOIP providers. - rmclaren
  • Minor : (Jul23/07) Phone volume changes randomly Using PCMU, HW 0.4. While in call, person on the recieving end will hear my voice get spontaneously louder and softer. I believe it's from the new echo canceler
- ninthclowd
    • Note : (July26/07) G726 occasionally sounds quite harsh. This may be the same issue. Not sure about the other codecs. - chewi
  • Minor : (Jul22/07) Some language translations are not viewing menu properly (missing characters in menu).
    • Slovak language is affected. - mnam
  • Note : (July20/07) Receptionist (with extension unit) at 75 phone location reports this is "Perfect".- diver
  • Minor : (Aug10/07) My call history time stamps are off by -1 hour. The time display is set to the correct time. Anybody else seeing this? - Shane Steinbeck
    • Note : (Aug13/07) yup, mine too - olly
  • Note : (Aug13/07) It would be handy if the status page showed the time/date on the phone so that these settings could be checked via the Web interface without having to visit the phone or asking someone to look.- Linker3000
  • Minor : (Aug13/07) The LCD backlight turns itself off during calls - I'm sure that in previous version, the light was illuminated all through the call. - olly
  • Minor : (Sep 4/07) Time of call does not include DST (daylight-saving time) - When you check call history, the time of call is displayed wrong. - bad2Dbone
  • Minor : (Sep 4/07) Phone keeps restarting with language pack if you are using provisioning with preseted language (tested with german) - bad2Dbone
  • Minor : (Sep 4/07) Noise in headset is much lower than in previous version, but it's still there. (professional headset with amplifier) - bad2Dbone
    • Note : (11.9.07) Noise appears sometimes after hanging up. It lasts for a while- bad2Dbone
  • Major : (11.9.07) Transfer sometimes doesn't work A calls B. B answers, picks new line and calls C. Than B picks back line with A, presses TRNF and line vith C. C hears silence, and cann't be picked up again - bad2Dbone







Firmware Notes 1.1.4.17:

This firmware can be downloaded from http://www.grandstreamsucks.com or http://www.idlex.net/?p=10

Build 1.1.4.17 (07/13/2007)

BugFixes:

  • Fixed LCD backlight switched off at inconsistent interval after off hook
  • Fixed GXP2000/2020 do not prompt user in dialing screen if account is not registered
  • Fixed incoming direct IP calling will not work if no accounts are configured
  • Fixed GXP2000/BT200 handset overdriving in TX path
New Features/Changes:
  • Changed we will place the phone on-hook after transfer is completed

  • Major : (Jul20/07) Breaks BLF light on GXP2000 BLF light is blinking after a call for up to 30 minutes.



Firmware Notes 1.1.4.16:


Build 1.1.4.16 (07/06/2007)

BugFixes:

  • Fixed GXP2000 noise on call hang up
  • Fixed GXP does not allow change network settings when "Lock Keypad Update" is set to Yes
  • Fixed GXP2020 screen does not clear cleanly when going offhook (some pixels left over from the date string at idle screen)
  • Fixed GXP2020 does not show SRTP icon when making an SRTP call
  • Fixed GXP2000 increased handset background noise in 1.1.4.14
  • Fixed BT200 does not play ring tone after call-waiting came in and onhook
  • Added BT200 onhook call-on-hold reminder message (port from BT100)
  • Fixed BT200 HOLD issue: hanging up the handset while having a call on HOLD will terminate the call
  • Adjusted GXP2020 GUI line info display for multiple call scenarios
  • Fixed handset ringer level too loud for GXP2000
  • Disabled silence suppression for G723/G729 to bypass the crash the problem
  • Fixed we will default to PCMU regardless of configured codec when incoming SDP contains video
  • Added option to disable use of multiple media attribute in SDP to workaround some platforms not supporting it. Provisioning parameters P137/487/587/687/1787/1887, takes immediate effect without reboot.
  • Fixed GXP2020 does not show "Sync Phonebook XMLâ?¦"
  • Fixed we use ";user=phone" in BLF SUBSCRIBEs even when "User is phone" is not selected
  • Fixed we will only update the first BLF/Presence status
  • Fixed GXP2000 high pitch sound when offhook using the handset
  • Fixed we do not honor session-timer refresher party when other the parameter contains a space between semicolon and "refresher=" parameter
  • Added support for Nortel MCS server-side conference
  • Added provisioning status on BT200
  • Added GXP offhook auto dial
  • Added support for BT200 provisioning display (Bugzilla #748)
  • Added support for new factory parameter section (to be removed for external release note)

  • Major : (Jul11/07) Line remains off hook after transfer After successfully transferring a call and hanging up, line1 remains illuminated red and off the hook. (HV .4). This is a real annoying bug. - JoeM <- FIXED IN 1.1.4.17 (iDLEx)

Firmware Notes 1.1.4.14:

Although not available as an official download as of Jun26/07, Grandstream support is mailing this firmware to people who request it.
This firmware can be downloaded from http://www.grandstreamsucks.com/ No pun intended!
Changelog 1.1.4.14:

Bug Fixes:

  • Fixed SRTP broken in 1.1.2.1
  • Fixed SRTP sequence number wrap around bug
  • Fixed we include "application/xpidf+xml" in Accept header for SUBSCRIBE
  • Fixed GXP-2000 Accounts 1 and 4 "SIP T1 Timeout" last option incorrectly labeled as "1 sec" instead of "2 sec"
  • Fixed we stuck at provisioning when receives a 200 OK with 0 content-length
  • Fixed DoS issue by WWW-Authenticate header
  • Fixed a crash issue with 489 Bad Request caused by Proxy-Authenticate header
  • Fixed we cannot parse DST string via configuration file correctly
  • Fixed ring tone download goes to incorrect TFTP server
  • Fixed we still claim an IP address after receiving DHCP NAK
  • Fixed description for "Enable Call Features" in web UI
  • Fixed we incorrectly formed the qop param without the quotation
  • Fixed we incorrectly parsed the nonce param
  • Fixed we incorrectly handled presence NOTIFY
  • Fixed we incorrectly handled dialog NOTIFY
  • Fixed we incorrectly responds 481 to refer NOTIFY when BYE arrives first
  • Fixed we accept broadcast SIP messages
  • Fixed phonebook download account index incorrect
  • Fixed GXP-2000 crash with EXT board
  • Fixed GXP-2020 always sends signal=2 in DTMF via SIP INFO regardless of the actual DTMF
  • Fixed the in-call timer does not tick when a call is on MUTE
  • Fixed the offhook status line disappears between DTMF digits and in-call
  • Turn MWI off when phone is offhook
  • Fixed phonebook download account index incorrect
  • Fixed CBCOM mode results in no audio
  • Fixed we display "HV: 0.4.255" on some old GXP-2000 hardware
  • Fixed Sylantro interop issue
  • Fixed we do not honor maddr parameter in SIP Contact header
  • Fixed we used the cached "realm", "nonce", or "opaque" parameter if they are 0 length
  • Fixed we do not display correct IP address after DHCP NAK
  • Fixed GXP2000 crash if you add entry to phonebook from call log and edit name/number field
  • Fixed we cannot authenticate using auth-int
  • Fixed a TCP interface bug and a HTTP server bug which should speed up web UI and fixes some display issues

New Features/Changes:

  • Improved audio quality
  • Changed description for "Enable Call Features" in web UI
  • Changed we use 408 instead of 487 for ringing-no-answer
  • Added wrap-around support for MENU UI
  • Added speed searching phonebook and other list box items
  • Added option to support mute speaker ring in headset mode (P336) --Note that this option does not exist for GXP-2000 HW 1.0
  • Added we will clear new missed calls display message after viewing Missed Calls menu without visiting details of each entry
  • Support for concurrent multiple DTMF schemes
  • Changed: We will bypass unregister and move forward to register if we received a non 2xx final response for the the (un)-REGISTER
  • Added we will clear new missed calls display message after viewing Missed Calls menu without visiting details of each entry
  • In web UI, the "if set to Yes, "#" will be function as the "(Re-)Dial" Key" is removed
  • Add display "Preparing to write" and "This may take a while" between the period of firmware download complete and firmware flashing begins
  • Disabled iLBC (iLBC is not working well and will be fixed later)
  • Added support for XML encoded in UTF-16
  • New dialing string display-old scheme will only show the last 11 digits dialed; new scheme will automatically scale down to a smaller font to display up to the last 42 dialed digits. Old scheme does not show line and account info while dialing digits, new scheme will.
  • New centralized GUI control allowing showing multiple call information simultaneously including:
  • Still show account information while dialing (previous version only show dialed digits)
  • When 2 calls are present, we will split screen in half vertically (3 lines per call) and display both calls
  • When 3 calls are present, we will split screen in 3 sections vertically (2 lines per call)
  • When 4 or more calls are present, we will split screen in multiple sections (1 line per call) and display as many calls as we can. If there is an active call, that call will occupy 2 lines.
  • Select scenarios will occupy the entire screen for the active call (such as when TRANSFER key is pressed, or when CONFERENCE key is pressed)
  • When TRANSFER key is pressed you will still see the current calling info along with the prompt or the transfer target number you are dialing
  • When CONFERENCE key is pressed you will still see the current calling info along with the prompt
  • When call forward are requested (such as *73), you will still see the account information along with a prompt and the forward target number as you enter

  • Upgrading to this firmware can take several reboots - be patient and wait for them to all be completed before logging back in to your phone/s.

  • Major : (Jun30/07) Incoming call on second line - no voice after pickup call (headset or speaker), first line is OK Incoming call on first line worked well, the only way hot to enable sound on second line afte pickup call is to hold the line and resume call, before that there is no sound on both calling parties . - Pteppic

  • Major : (Jun27/07) Speaker always on Speaker hisses even if no call active (HW v2.0). - Tigs

  • Major : (Jun27/07) Line remains off hook after transfer After successfully transfering a call and hanging up, line1 remains illuminated red and off the hook. This is unacceptable! - JoeM
  • Note : (Jun28/07) confirmed. HV=0.4 - ampster

  • Major : (Jun26/07) Breaks HOLD button on BT200 It has been reported in the Trixbox forums that this firmware stops the HOLD button from working on the BT200 model. GXP-2000 not affected. - Linker3000

  • Question : (Jun26/07) I'd like to know, whether the new firmware policy will be to issue upgrades only upon request, whether this firmware is considered stable or not (I don't get the point in the question marks behind the word stable... shall this be considered 'beta, although (of course) it's hoped to be stable'?). and whether there will be new official stable images distributed via the official site (as opposed to 'only upon request')? - kodomo
    • Answer: (Jun29/07) One of the engineers at Grandstream contacted me and asked me to post this response to some of the questions about the firmware and these pre-beta versions we keep seeing. I think he addressed this situation very well, and I will summarize by saying: "Of course it's broken and there are bugs with it. It's pre-beta. Grandstream appreciates all of the feedback on problems, but it does not make sense to expect the pre-beta firmware to be stable or even have all of the functionality one would expect from a stable release or full beta release candidate." Upgrading is a one way street, and they warn time and again against doing it in production environments before the firmware is released as stable. Anyway...back to the engineer's response: - thetatag
      • We did a lot of hard work to get to 1.1.4.14, especially with the audio fix. That is not an easy one given that we cannot change the hardware case to maximize the physical sound path. Initial internally rolling testing seemed fine, but we did find some test phones needed some time to get stabilized (they will freeze after f/w upgrade) and the chirp sound in on/off the handset. Also the BT200 HOLD key is broken (although the FLASH key can do a similar thing as HOLD). That is why we halted it in pre-beta and did not even go to beta as we try to nail down these issues before publishing into beta, because the update is a one way train.

  • Note : (Jun28/07) Yes, I have some background hissing (white noise) but the firmware is so much better than any before. THANKS.- richard
  • Note : (Jun28/07) I have always had some slight but barely noticeable hiss (white noise). HV=0.4 - ampster

  • Note : (Jun28/07) some improvement in speaker phone also. but there is an initial burst in speaker mode. HV=0.4 - ampster

  • Note : (Jun28/07) Overall, the firmware keeps on improving. HV=0.4 - ampster
  • Note : (July04/07) First version in a year or more that DOESN'T BLANK THE SCREEN (HV0.3). That is very good, it has worked very well for 2 days since installing the new FW. Less good though - POSSIBLE WARNING - My phone died today, it is now a flashing brick. I can't rule out that it has to do with this new firmware that has been on for 2 days.- carlotto



Firmware Notes ("Beta" 1.1.1.13):


Changelog (1.1.1.13)
  • Audio adjustments
  • Fixed some v 0.3 and 0.4 blank LCD issue
  • Added "application/dialog-info+xml" in the Accept header of 415 response
  • Added Syslog for SIP dialog matching result
  • Fixed init problem causing the GXP-2000 v1.1 bootup problem

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.1.13.pdf" file included with the firmware. - flu

Bugs / Tweaks (1.1.1.13):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR: (Sep29/06) SCREEN BLANKING Took about 15 mins for it to blank. Come on Grandstream, this is getting very very tiresome. Carried over from the mists of ancient time, onwards. - mattb
    • NOTE: (Oct01/06) Hey mattb, have you tired another power adaptor? I had one of my phones doing crazy things and found that the power adaptor was giving out strange voltages, changed out and works like a dream. - jase495

  • MINOR: (Oct02/06) GUI still is very ugly and hard to use. Please consider the changes your users suggested below (search for "low usability" on this page). - job

  • MINOR: (Oct02/06) The naming of line 10 and 11 (":" and ";", respectively) is unexpected enough to be called a bug. If it's impossible to call them "10" and "11", go for "A" and "B" or something. - job

  • MINOR: (Mar29/06) Numbers off-screen. When entering long phone numbers (i.e. for international calls) the phone number goes off-screen as it advances, rather than moving up to the upper line as it used to. Carried over from 1.1.0.13 onwards. - Mike

  • MINOR: (Sep24/06) State Bugs. Loopback call (dial phone from itself, via asterisk) drops out on answer (as expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up/down arrow adjusts the volume instead of going to phonebook. Carried over from 1.1.0.13 onwards. -Jedi98

  • MINOR: (Oct05/06) Call timer on LCD appears to wrap around, presumably at 12 hours. Noticed this while testing phone stability. - edgar

  • Question: (Sep30/06) Audio quality. Is the audio any better in this release? I'm still on 1.1.0.11 and dont feel like risking it unless audio is OK. - falle
    • A: (Oct01/06) IMO Audio was OK upto and inc. 1.1.1.7 and then wen muffled. At 1.1.1.13, g711 (PCM) is better than 1.1.1.12 but not perfect. g729 remains muffled by comarison to 1.1.1.7. Other users have had different audio issues but I am not sure if these have now been solved. I would advise holding off if possible, unless you have other bugs you need fixed, since you cannot downgrade. .13 is too new for all the bugs to be in yet. - jedi98
    • A: (Oct01/06) Yes, I'd say it's "different". It's much better than before, but the sound was clearer in 1.1.0. - job

  • Note : (Oct05/06) Stability. I have found that 1.1.1.13 seems to have fixed stability issues I saw with some of my phones on 1.1.1.9. These phones would often hang at boot and sometimes freeze both on and off calls. So far, these upgraded phones have not frozen on/off calls or hung at boot. I will probably be upgrading all my phones to 1.1.13 from 1.1.1.9 after testing it on a few user desktops for a while. - edgar




Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)

  • MAJOR: (Oct 2/06) Sidecar stopped working in v1.1.1.13 (was working with v1.1.1.12), red light just flashes fast (uninitialized). Not using AC adapter to power sidecar. - andrew4455
    • NOTE: (Oct03/06) Confirmed, Red flashing light, does not initialize. Rebooting from GUI crashes the phone. - joekane


  • MINOR: (Oct 2/06) I called tech support about the audio quality when using the sidecar, and was told to NOT use the ac adapters with the sidecar(s) with the GS 2000 hardware version at 1.1. I took off the adapters, and the audio quality is better, but not quite good enough. I am on .12, and waiting to see if there are other issues with .13. Current problem is that voice is choppy at times.




Firmware Notes ("Beta" 1.1.1.12):


(Sep22/06): Currently available from Grandstream BETATEST site, 1.1.1.12 Firmware, 1.1.1.12 Release Notes.

Changelog (1.1.1.12)
  • No change for GXP-2000 from 1.1.1.11
  • Disable the EMIF optimization which caused some BT-200 to fail to upgrade


Bugs / Tweaks (1.1.1.12):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR (Sep24/06) Audio quality on speaker and handset is still muffled (by comparison to 1.1.1.7). Codecs tested were 711 & 729. High frequencies are cut and distorted, it sounds like the phone has a slightly stuffed up nose (A phone virus maybe? ;)) Carried over from 1.1.1.9 onwards. - jedi98

  • MAJOR (Sep25/06) Unable to Register via LAN to LAN. Example, the phone is on my private lan (192.168.11.0/24) and my default gateway has a VPN session to my office LAN (192.168.44.0/24). The Grandstream GXP-2000 registers fine with my local asterisk server (192.168.11.10) but won't register with my office server (192.168.44.2). Checking tcpdump/stats the phone doesn't even send the sip to the default gateway, perhaps its not sending to d/g because it sees the IP as in RFC1918 and is hard-coded in the firmware that all rfc1918 doesn't need to go via the gateway? (just a guess!) Bug is present in ALL previous firmware releases tested, at least back to 1.1.0.16 - andyb2000
    • NOTE: (Sep25/06) I have similar config but I am NOT getting the problem. I register acct 1 with a local * server and acct 4 with an * server on another subnet in vpn. Subnets are 192.168.1.0/24 and 192.168.11.0/24. VPN is done by routers. So there must be somthing differing in our configs somewhere. - jedi98
    • NOTE: (Sep25/06) Interesting, would you take a look at my profile page here and get in touch? Many thanks - andyb2000

  • MINOR: (Sep24/06) State Bugs. Loopback call (dial phone from itself, via asterisk) drops out on answer (as expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up/down arrow adjusts the volume instead of going to phonebook. Carried over from 1.1.0.13 onwards. -Jedi98

  • MINOR (Sep26/06) Speakerphone mic too sensitive. In 1.1.1.10, they increased the speakerphone mic gain, and now it is much too sensitive. It picks up all the background noise, and it sounds (to the other side) like you are in a train station. Ideally, the gain should be adjustable in config. - bcheath

Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)

  • MAJOR (Sep26/06) Extension Unit hanging In 1.1.1.12, Using asterisk 1.2.7.1-BRIstuffed , Maybe its the asterisk version im using but the extension unit just hangs after about 10 minutes of operation. A reboot fixes it. - joekane

  • MAJOR (Sep26/06) Paging Issue In 1.1.1.12, Using asterisk 1.2.7.1-BRIstuffed , When paging to 40 GXP's alot of the lights still stay on even tho remote disconnect is enabled on all phones. Reboot fixes. (Phones on 1.1.0.16)??? - joekane

(In addition to adding user feedback here, please copy items from the previous bug/tweak list to this list if you discover that they are still issues and note that you have done so! Thank you!)



Firmware Notes ("Beta" 1.1.1.11):


NOTE: (Sep16/06) Firmware 1.1.1.11 has been pulled from beta site. Kurgan


NOTE: (Sep14/06) The filename suggests this is a Release rather than a Beta although it appears only on the Betatest site currently. - Channel-Two
  • (Sep14/06) The betas always say Release_GXP2000-BT200_version, I don't understand how this one is different. I think they're only considered stable when they are in the DOWNLOAD directory instead of BETATEST. - Shane Steinbeck
  • (Sep15/06) Okay fair enough. - Channel-Two

Changelog (1.1.1.11)
Fixed Phonebook problem again?

Bugs / Tweaks (1.1.1.11):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR (Sep15/06) Audio quality on speaker and handset is still muffled (by comparison to 1.1.1.7). Codecs tested were 711 & 729. High frequencies are cut and distorted, it sounds like the phone has a slightly stuffed up nose (A phone virus maybe? ;)) Raised to major because audio is the phone's primary function!. - jedi98

  • MAJOR: (Sep15/06) Display still blanks after some time. MAC 00.0B.82.03.XX.XX, H/W v0.3. Accessing the phonebook will cause it to come back, unfortuantly picking up the handset won't. Sometimes triggered by a call. - jedi98
    • NOTE: (Sep15/06) I confirm this, again. BTW, this is my first and last phone from Grandstream. - julianjm
    • NOTE: (Sep15/06) Confirmed - mattb
    • NOTE: (Sep16/06) I have never experienced display blanking, I have about 20 phones, the oldest has MAC 00.0b.82.05.aa.02. It definitely means that it's a hardware issue, probably in some specific HW revisions. Maybe GS should clearly state what HW revisions are faulty, and stop trying (but did they even actually tried, or are we just hoping that the are trying?) to fix this bug in firmware. - Kurgan
    • NOTE: (Sep16/06) The point to remember here is that when these phones came out with 1.0.1.9 firmware there was no problem with the disp. at all. But some functions were to be added with later firmware, so the phones were essentially sold with functions that required new fw to use them. Upto at least 1.0.1.13 there was no problem here, but those who went beyond cannot downgrade and the .03 rev phones were on sale until very recently. - jedi98
    • NOTE: (Sep19/06) Strangely enough i have 3 of my MAC 00.0B.82.03.XX.XX phones running 1.1.1.11 and none of them are having blanking problems (that we have noticed), every firmware that i have tried until now has caused the problems though... - SoloFlyer
    • NOTE: (Sep19/06) Scratch that.. my phone just blanked twice within about 15 mins... - SoloFlyer
    • NOTE: (Sep20/06) Ok, so it seems that newer firmwares trigger the display blanking on phones that previously worked well. I supposed that on older phones any firmware after the legacy 1.0.0.x caused the problem, but I seem to be wrong. - Kurgan

  • MINOR: (Sep15/06) State Bugs. Loopback call (dial phone from itself, via asterisk) drops out on answer (as expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up/down arrow adjusts the volume instead of going to phonebook. -Jedi98

Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)


Please sign your posts and also enter a comment for the wiki history in the 'comment' box! Pages of unknown edits are not helpful!

(In addition to adding user feedback here, please copy items from the previous bug/tweak list to this list if you discover that they are still issues and note that you have done so! Thank you!)






Firmware Notes ("Beta" 1.1.1.10):


Changelog (1.1.1.10)

Release Note
Firmware Version 1.1.1.10
September 13, 2006
Firmware 1.1.1.10 has major changes (compare with 1.1.0.x Releases) which may requires firmware to be downloaded twice(and reboot itself), and it can not be downgraded to previous version.
Make sure all the files that come with
Release_GXP2000-BT200_1.1.1.10.zip is unzipped into the TFTP or HTTP server.
For any firmware upgrade from 1.0.1.x or 1.0.2.x, please refer to previous release note and firmware and upgrade them to 1.1.0.16 first.
===============================================================
Product: GXP2000
Date: 2006-09-13
Release items: boot55a.bin 1.1.1.2
gxp2000a.bin 1.1.1.10
Previous release: boot55a.bin 1.1.1.1
gxp2000a.bin 1.1.1.9

Release Note for GXP-2000 and BT200
Build 1.1.1.10 9/12/2006

  • Fixed GXP-2000 crashes when a very large phonebook file is downloaded
  • Fixed GXP-2000 crashes when "Remove Manually-edited entries on Download" is set to Yes
  • Fixed GXP-2000 Name is not displayed for multi-functional keys on EXT
  • Fixed SIP stack incorrectly parsed "CT" header
  • Turn on this option by provision parameter P339 (1: use Account Name, 0: use date). First 12 digits are displayed, aligned to center (odd length 1 slot to the right), undisplayable characters will be blank
  • Support for 5 provision attempts
  • Fixed GXP-2000 take account 1 information when replying the missed call for account 2
  • Fixed GXP-2000 Speaker mode is triggered on 2nd call when the first call ended
  • Fixed audio quality degraded when call-waiting tone is been played
  • Fixed we cannot correctly parse incoming SIP messages with Contact headers that come without a username part causing some Broadsoft test cases to fail
  • Support for Broadsoft click-to-hold (Allow-Events: hold)
  • Fixed G.723 6.3kbps decoder does not work, web UI enabled this option
  • Increased speakerphone mic gain by 7.5db

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.1.10.pdf" file on the BETATEST download site. BT200 Specifics removed - Shane Steinbeck

Bugs / Tweaks (1.1.1.10):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR: (Sep14/06) Speakerphone: Sounds terrible "like you are in the twilight zone." - Shane Steinbeck
  • MAJOR: (Sep14/06) Screen: Screen still blanks after a few minutes... yawn.... - mattb

Please sign your posts and also enter a comment for the wiki history in the 'comment' box! Pages of unknown edits are not helpful!

(In addition to adding user feedback here, please copy items from the previous bug/tweak list to this list if you discover that they are still issues and note that you have done so! Thank you!)



Firmware Notes ("Beta" 1.1.1.9):


NOTE: (Aug14/06) NEW file appears to be available http://www.grandstream.com/BETATEST/GXP2000_BT200/Release_GXP2000-BT200_R1.1.1.9.zip though it contains boot55b.bin bt200b.bin gxp2000b.bin and my phone only tries to get the a version of these files. - Andy
NOTE: (Aug14/06) The missing "a" versions have been added by Grandstream today by updating the firmware file. I was able to upgrade to 1.1.1.9. - kam
NOTE: (Aug14/06) I had no problem with upgrade to 1.1.1.9 with "b" on my GXP2000 with 1.1.1.7 previously installed. - kondor

Changelog (1.1.1.9)

  • Improved audio quality
  • Modified memory management for iXML parser. This should resolve the freeze on downloading 50-record phonebook XML problem
  • Fixed several GUI menu bugs Fixed GXP-2000 does not save after more than 30 extension entries
  • Fixed GXP-2000 does not store UserID for KEY36
  • Fixed GXP-2000 cannot answer incoming call when in the SIP proxy edit screen
  • Fixed the screen XML '$d' variable does not display correctly
  • Fixed BLF does not activate speed dial when BLF party is in use
  • Fixed GXP-2000 continue to ring when BYE is received for early dialog
  • Fixed we do not clean out the call properly when terminating a call due to SRTP not enforced
  • Fixed we will perform firmware upgrade even if configured not to when DNS query for config server failed/we query "0.0.0.0" when configured such in firmware/config servers
  • Fixed a memory-leak issue that is only exposed by how GXP-2000 handles attended transfer (does not apply to other products)
  • Fixed GXP-2000 does not place transferee on hold when attempting to transfer
  • Extend the original "Disable missed-calls" feature to allow a new mode to disable all call-logs on a per-account basis. P182/442/542/642: old values (0/1), new values (0/1/2) where 2 means disable call-log.
  • Fixed both GXP-2000 and BT-200 turns speaker on when MSG key is pressed even when no voicemail user ID is configured
  • Fixed a potential crash if a NOTIFY with bad dialog XML
  • Added a memory debug feature: on right-top corner current memory status is displayed in lieu of time (or date, if reversed) in the format of x/y where x is the current usage and y is the peak usage

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.1.9.pdf" file included with the firmware. BT200 specific changes were removed. - flu

Bugs / Tweaks (1.1.1.9):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR: (12 Sept 2006) Voice Mail UserID: When this is set, the phone is unstable and locks between 5 and 20 minutes. This value was set to the shortcode for checking VM. If this is incorect usage, the phone should fail gracefully.
    • NOTE: (Sep19/06) I haven't seen this behaviour with about 45 running GXP2000 phones - edgar

  • MAJOR: (Sep05/06) RTP Decode: When the GXP-2000 receives an RTP packet, it assumes that the packet is encoded with the same codec that the GXP is currently sending. This need not be the case. The phone should look at the payload type byte in the packet to decide which decoder to use. The result is one way audio even though the RTP stream can be seen to be flowing in both directions correctly. - marvy

  • MAJOR: (Aug31/06) Call Waiting: When call waiting tone is enabled and a second call comes in, the audio of the call you are on is effectively muted permanently. - forrestc

  • MAJOR: (Aug15/06) Phone Book: When attempting to update phone book and having the option "Remove Manually-edited entries on Download" set to Yes, the phone still freezes (clock stops, doesn't ring on call, totally unresponsive until power cycle) with "Sync Phonebook XML... This may take a minute" on screen. - acabtp

  • MAJOR: (Aug16/06) Lockups: Seems to randomly lock, I'm using BLF so it may be when states of those change, however after about 10mins the phone totally locks up and needs power reset - Andy
    • NOTE: (Aug14/06) You will want check your network for odd behavior. Most of the phone lockups I found were due to bad patch cords or misbehaving equipment on the network. My T1 at one point was the culprit. Due to a bad smart jack the te110p was sending out false data to certian phones. Another cause is the syslog. With the syslog enabled on too many phones our network took a huge hit and many phones acted up. - diver
    • NOTE: (Aug23/06) Not based on cabling, etc as those were already checked, but 24hrs later and a few reboots the problem has gone away, as has the web interface crash below - Andy
    • NOTE: (Aug31/06) I'm also seeing locks where you will be talking on a call and the phone simply locks up - no audio, can't hang up, etc. Needs power cycle to recover. I'm almost 100% sure there are no network issues causing this. - forrestc
    • NOTE: (Sep19/06) Seeing random lockups on a few phones, never while in use. These phones also tend to have problems coming up after reboot. Reflashing doesn't seem to fix so I am RMA'ing phones doing this. - edgar

  • MAJOR: (Sep07/06) Display still blanks after some time. MAC 00.0B.82.03.XX.XX, Pressing certain buttons on the hand set will cause it to come back, unfortuantly picking up the handset wont - SoloFlyer
    • NOTE: (09/08) Happens with MAC 00.0B.82.03.CC.2F. HW revision: 0.3. It seems, that 0.4 is OK - FESTR

  • MAJOR: (Aug16/06) Web Interface: Trying to access web interface, allows password/login but as pages start to load they freeze after perhaps one or two input boxes can be seen, phone then crashes, needs power reset. - Andy
    • NOTE: (Aug23/06) 24hrs later and a few reboots the problem has gone away, as above (lockups) - Andy

  • MAJOR: (Sep/06) Registration: Phones often do not reregister if they lose their connection to asterisk, for example if the server is rebooted. Often they also stop responding to pings and the web interface is inaccessible even if the phone is forced to reregister. I force this by running the ethernet loopback test for a few seconds after which the phone will reregister in about 30 seconds. This is a major inconvenience if you have to go to most of your phones and manually make them reregister! - edgar

  • MINOR (Sep19/06) Cannot access the web interface if phone is in use. You get a message saying the phone is busy. - edgar

  • MINOR (Aug15/06) Audio on speaker and handset is muffled by comparison to 1.1.1.7. Codecs tested were 711 & 729. I downgraded the phone to 1.1.1.7 and the audio was clear again. It sounds like the volume was boosted slightly at 1.1.1.9 but the high frequencies were cut or distorted. - jedi98
    • NOTE: (Aug24/06) I can confirm a quite bad sound (distortet and some freqs cut off). AGC on handset's speaker still gives bad volume changing. - fratzr

  • MINOR (Aug15/06) The audio crackles on g.726 codec (the same on 1.1.1.7 firmware). - Maxo

  • MINOR (Aug23/06) Every few mins using BLF I am getting the following in asterisk logs, and the phone doesn't update the BLF indicator for that SIP channel.
Incoming call: Got SIP response 415 "Unacceptable Content-Type" back from 192.168.2.2
(that IP being the phone that has the BLF indicators set)
- Andy

  • MINOR (Aug03/06) - BLF State Freezes For some reason BLF seems to work fine on my phones for a while, but then something causes all of the lights to freeze in whatever state they were in (i.e. ringing, idle, off hook) and they won't refresh until reboot. I just upgraded to asterisk 1.2.10 around the same time I upgraded my firmware. Prolly a dumb move, however I am getting no errors whatsoever from the asterisk logs. Anyone else having this problem? - ninthclowd
    • NOTE: (Aug08/06) I think I found out what is happening. Lets say there are two phones, A and B. Phone A has a BLF button setup for Phone B. Something then causes phone B to lose its registration until reboot <tangent> I have no idea why this is happening but it only started happening this firmware and should probably be listed as a major bug in itself as it will drop calls </tangent> The BLF state for Phone B on Phone A will freeze in whatever state it was in when it lost registration and won't clear until reboot. - ninthclowd
    • NOTE: (Sep10/06) I am having this problem with 1.1.1.9, but I don't think my problem is being caused by lost registrations. - vinceval

  • TWEAK: (Aug14/06) SIPS support: It's really nice to have SRTP now - however, with SDES the master password is sent unencrypted in the SIP stream AFAIK, so for real security it might be useless... Therefore the GXP-2000 should also support SIPS to make this solid (or is it already supported?) - Mirak

  • TWEAK: (Aug16/06) Phone Book: The numbering system in the phonebook has changed. In the previos version the phone lines numbering started with 0 and end winth 3. In the new firmware the lins are numbered as they are on the front panel, Line 1 is 1, Line 2 is 2, Line 3 is 3 and Line 4 is 4. I've made a XML file phonebook and I had to go +1 on every entry in <accountindex> </accountindex>. - kondor

  • TWEAK (Aug22/06) built in microphone When you use the handset and press the speaker button the microphone should not switch to the build in microphone but should stay in the handset. This function is used for letting others listen to a conversation without notice, when the handset is on hook the built in microphone should be used. Right now its not possible to let someone listen to a conversation without lower quality for the called party, even old ISDN phones have this feature.- Datu
    • AGREE WITH YOU (Sept16/06) This feature is very common and will arrange echo problem...- flo_turc

Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)

  • MAJOR: (Sep11/06) Sidecar: Buttonnames aren't displayed only the telephone number by pressing a speeddail key on the extension unit. The speeddial keys on the GXP-2000 does display the name and telephone number. - LJPYRO

  • MAJOR: (Aug16/06) Sidecar: Still doesnt work with using the web interface, buttons pased 18 dont light up *Asterisk BLF*. this is very disapointing Grandstream. Selling the hardware months before it can work, get it sorted will you - Joe

Please sign your posts and also enter a comment for the wiki history in the 'comment' box! Pages of unknown edits are not helpful!



(In addition to adding user feedback here, please copy items from the previous bug/tweak list to this list if you discover that they are still issues and note that you have done so! Thank you!)




Firmware Notes (Beta 1.1.1.7):


NOTE: (Aug11/06) The file has been pulled from the beta download site. - Shane Steinbeck

Currently available from Grandstream. This is also a browsable directory, more stuff available at Grandstream downloads or Grandstream Beta Site Home.

WARNING: This is a BETA version. It may have stability issues and is provided by Grandstream for testing purposes only. Do not use in a production environment untested.

Changelog (only GXP-2000 related)
Build 1.1.1.7

Fixed custom ring tone by Alert-Info fails
Added option to check incoming INVITE sip user ID
Fixed DTMF buffer not cleared when switching lines for unestablished dialogs
Support disable call-waiting tone
Add UCF (Unconditional Call Forward) icon on status line
Fixed high pitch done played when Call Forwards are enabled and disabled
Fixed user cannot enter * and # in phonebook entries. In addition, user can enter @ by using HOLD key in phonebook submenu
Fixed we crash on attended transfer on platforms that use To/From headers without square brackets
Fixed we still responds "recvonly" on un-hold SDP message
Fixed GXP-2000 ring tone change via keypad menu not effective after reboot
Added volume control is stored after reboot
Added Support for GXP-2000EXT keys in diagnostic mode
Disabled headset side tone
Fixed IP Fragmentation bug
Add Support for IM and screen XML feature (saving to flash)
Fixed we send NTP to wrong IP address
Added force LCD update on hook status change (this makes LCD GUI look more responsive when onhook)
Added customizable idle screen via downloading XML by HTTP/TFTP
Added support for SIP MESSAGE method (RFC 3428); stores up to 100 incoming IM messages, after that new messages are dropped
Added support for SIP PUBLISH method (RFC 3903)
Added support for SIP Presence package (RFC 3856, 3863) for use of 7 MFKs and GXP-2000EXT
Added support for SIP Dialog package (RFC 4235)
Added support for SRTP by SDES
Fixed GXP-2000 crashes when speed dial user ID contains '@'
Fixed the clock on the right top corner displays incorrectly if switches from 12hour display to 24hour display.
Added support for G.726 codec
Added support for GXP-2000EXT console.
Added support for anonymous call using privacy header
Added support for downloadable phonebook

  • IMPROVED: Speakerphone volume has been improved. This is a work-in-progress as Grandstream is working hard to preserve accoustic quality while increasing the volume of the speakerphone. It hasn't "arrived" yet, but it's moving in the right direction. - thetatag
  • IMPROVED: Forced screen refreshes on certain events takes care of many of the screen problems with older hardware. - thetatag
  • IMPROVED: Handset, spealerphone, and headset volume are now all independent.- thetatag

Bugs / Tweaks (1.1.1.7):

Please sign your posts and also enter a comment for the wiki history in the 'comment' box! Pages of unknown edits are not helpful!
(Jul28/06) Could you please describe what a useful comment would contain? thanks. - Anthony

(In addition to adding user feedback here, please copy items from the previous bug/tweak list to this list if you discover that they are still issues and note that you have done so! Thank you!)

  • MAJOR WARNING: (Jul28/06) There is no Turning Back I tried going back to firmware 1.1.0.16 through tftp and reboot, no luck. So I tried a full reset using the menu and the MAC address. The phones still have firmware 1.1.1.7 loaded. duh. I was hoping to recover the audio quality of the older firmware, no such luck. - Anthony
    • NOTE: (Jul29/06) Also got caught out. Tried to downgrade with HTTP. But, I'm now so used to getting caught out by these betas that I gave up complaining! Plea to GS: If you cannot provide a method for downgrade then CAN YOU PLEASE show a big warning with the notes for a non-reversable releases (BETA or otherwise). - jedi98
    • NOTE: (Aug08/06) /Agree with the Jedi - ninthclowd

Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)

  • MAJOR: (Jul16/06) Any entries after button 18 dont save, they just disappear. Same goes when 2 extension units are connected. - joekane
    • NOTE: (Jul16/06) By using the Windows or Linux config file creator you can put these values in the fields. They will still disappear when you use the web interface after you download the cfg. - diver

  • MINOR (Aug12/06) Some lights on the extension unit do not consistently work properly. Several phone indicator lignts (3 out of 45) will go blank instead of Red when a call is picked up. These phone indicators will show blinking Red when being rung, so it is not the LED. One phone at position 15 does not show status at all. All phones were setup with the same configuration using a template. - diver

  • MAJOR: (Jul16/06) Red lights stay flashing when paging extensions setup as "Asterisk BLF". - joekane

Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR: (Jul16/06) Display still blanks after some time. MAC 00.0B.82.03.XX.XX - julianjm
    • NOTE: (Jul18/06) It's funny because I've never seen my screen blank before but with this firmware a few of my phones did, however pressing the MUTE button fixed it as described in the previous firmware - ninthclowd
    • NOTE: (Jul19/06) One of my screens flipped upside down and had to be restarted. - diver
    • NOTE: (Jul20/06) On my phone blanking never went away. However, seems more common with this version. I've seen vertical scroll too, which then fixed itself immediately, and also blank when idle that refreshed on pressing '#'. Are we looking at a hw problem being worked around by sw or a recurring bug? - jedi98

  • MAJOR: (Jul16/06) Phone Book: When attempting to update phone book via TFTP or HTTP with XML file of format specified in 1.1.1.7 user's guide, phone freezes (clock stops, doesn't ring on call, totally unresponsive until power cycle) with "Sync Phonebook XML... This may take a minute" on screen. - acabtp
    • NOTE: (Jul16/06) I have the same issue, but it only seems to hang with more than 15 address book entries in the XML file. - gammacoder
    • NOTE: (Jul16/06) I am experiencing the problem with any sized XML file. If someone has an XML file that worked, please post it as an attachment so others may test. - acabtp
    • NOTE: (Jul16/06) Problem occurs with both PC and Unix formatted text files. Server logs indicate that the .xml file is successfully downloaded before the phone freezes. - acabtp
    • NOTE: (Jul16/06) The example in the manual worked for me. Copy and paste. Find it and a dynamic example at http://www.voip-info.org/wiki/view/GXP-2000+XML+Phonebook - Shane Steinbeck
    • NOTE: (Jul18/06) I couldn't get it do download without freezing untill I switched off the "Delete Manual Entries" now it's working a treat - Peter Almgill
    • NOTE: (Aug02/06) For me, anything between 15 and 20 entries is hit and miss. Sometimes it works, sometimes it freezes. Anything above 20 entries freezes every time. This is irrespective of whether "Delete Manual Entries" is enabled or disabled. - Channel-Two

  • MAJOR: (Jul16/06) The audio quality on all codecs is much worse with this firmware. The audio crackles (sounds like overmodulation), and adjusting both the codec and the TX frames per packet did not resolve the problem. Hopefully I'm missing an option that needs to be re-configured. - mansing23
    • NOTE: (Jul18/06) Agreed. I've noticed it as well - ninthclowd
    • NOTE: (Jul19/06) ditto - Anthony


  • MAJOR: (Jul20/06) Automatic Gain Control. Perhaps as an attempt to improve audio volume, Grandstream introduced an Automatic Gain Control (AGC) system. A distinct difference can be heard between older versions (more than a couple versions ago) and the newer ones, particularly that nearly every word you hear on the phone starts out quiet and gets loud towards the end. Short words like "no" spoken apart from any other words get nearly lost in the quiet. It is a poor implementation of an AGC, and even if it was not, any AGC that's noticable should be optional. This is not an acceptable approach to increasing the volume of the GXP-2000. I believe this qualifies as a major bug as this prevents this firmware version from being useful in a production environment.- thetatag
    • NOTE: (Jul20/06) First time I heard this effect I thought it was due to some heavy drinking the night before and ignored it lol. But you're absolutely right, this is definately annoying, and I don't believe it is worth keeping it, as the problems with the speakerphone remain. Speaking of the speaker phone... is it getting worse or is it just me? - ninthclowd

  • MAJOR: (Aug02/06) Vocoder Error. When "SRTP Mode:" is set to "Enabled but not forced" and you try to make an outgoing call you get an error on the screen saying "488 NOT ACCEPTABLE. Try a different vocoder." Seems to happen no matter which codec you choose. If it is not forced should it not fall back to RTP and allow the call to proceed. - naturalblue

  • MAJOR (Jul24/06) - Custom Idle Screen:Using custom bitmap with offset causes phone to crash. Suspect chunks of memory are being overwritten by the offset routine. The problem reoccurs every reboot when the phone attempts to load the Custom SCR. To remedy this, I had to reboot the phone with the network cable disconnected, and clear the custom screen from the preferences menu before it could be loaded. - acabtp
    • NOTE: (Jul31/06) I am using an offset of 3 for X and 18 for Y with no problems. Are you using a full screen image? - ninthclowd

  • MINOR (Jul24/06) - Custom Idle Screen:Random corruption on screen when using custom XML file. Noticable as dots or lines on the extreme left and right of text blocks. - acabtp
    • NOTE: (Jul25/06) This might be hardware or a problem with the way you encoded the image. I have 30+ phones that are displaying the same image with no corruption. Maybe its possibly a problem with the XML file corrupting during download? - ninthclowd
      • NOTE: (Jul28/06) There is no custom bitmap, only text fields in the configuration files I am using. I will post a picture on Monday. - acabtp

  • MINOR (Jul24/06) - Custom Idle Screen:Using the $d variable reference causes the phone to display only the last digit of the day of the month, and the rest of the <DisplayStr> after the $d is truncated. - acabtp

  • MINOR (Jul24/06) - Custom Idle Screen:The phone seems to always evaluate the "a1reg" as true, resulting in any <DisplayStr> that have "a1reg=false" never being displayed, and <DisplayStr> that do not have "a1reg=false" always being displayed, regardless of the Line 1 registration state. - acabtp

  • MINOR (Jul25/06) -Jitter buffer:I noticed the receive audio latency grows to around 100-200 ms (over approx 1-2 minutes) and never shrinks. There is no significant jitter on my system as everything is on the same LAN. I think something in the jitter buffer in the phone is not correct as I don't see this behaviour on other SIP devices on the same network.

  • MINOR: (Jul16/06) * AND ** do not work for speed dial. I use * for intercom (yes I know you can train the users to just press *, but they actually want a button labeled) and ** for Transfer to VM. If I program an extension with the * it works; but either by themselves just resets the state of the phone and I have seen the screen flash ** for the speed dial content of **. If dial manually it works. Any one know what the phone sends when you press a speed dial? - diver
    • NOTE: (Jul18/06) I might be wrong, but I'm pretty sure that if you put an extension in the speeddial buttons it gets prefaced with two asterisks. I wonder if when your putting the * into the speeddial it's getting translated to ***? If that was the case it would be possible to just add a *** and **** extension in the asterisk dialplan that rerouted to the original context. Just a thought O.o - ninthclowd
    • NOTE: (Aug11/06) I have worked on this some more and have determined the problem lies with the speed dial function. This did work for one week before upgrading to .13 or .16 (I don't remember now which one). The latest firmware sends the call (which is appropriate when you think about it). What I want to be able to do is have a field in the web/cfg that toggles whether or not the call is sent immediately after the speed dial button is pressed. I want to be able to say dial this number (in my case ** or *) then wait for user input before the call is sent. Yes, users should be able to press two buttons to send a call to voicemail, but it is very nice to have a button that is labeled "XFR/VM". - diver

  • MINOR: GUI-style menu system still has low usability. Colors are inverted for no obious reason. "Del"-key is delete and not backspace, which would be expected. No menu shortcuts on keypad. No keys which represent "ok" and "cancel" (have to navigate with arrow keys). - job
    • NOTE: (Feb07/06) Perhaps the up and down silkscreened/segmented arrows at the right of the display might be useful here? IMHO, simplify the GUI as much as possible, expand the text/list area to the whole screen width, and by dumping some UI you can get another line of text vertically. -Helix
    • NOTE: (Feb07/06) IMO it should be clean, uncluttered, no need to emulate windows, no need for scroll bar arrows. -Jedi98
    • NOTE: (Feb11/06) Agreed. There's no need for buttons and scroll bars. Something like this would be ideal for the menus � the user can select an item just by pressing a number (no need to use up/down, and options off the screen can be accessed w/o scrolling). The down arrow to the right of the LCD is on to indicate there is more beyond what is on the screen. Also, since the left arrow button goes back to the previous menu, the right arrow button should function like the select button. Similarly, when in a configuration screen, a simple layout like this for a phone book entry would be much easier to navigate. Up/Down Chooses a field. Press the Select or Right button to edit that field. When editing, the cursor is either an underscore, or maybe reverse text (better than the current �|� type cursor that makes the text next to it unreadable). Delete/Mute should act as backspace, just like it does when dialing a number. Pressing Select again finishes editing that field, and the user can use up/down/numbers to select the next field. For a toggle field, select toggles the value. -Ted
    • NOTE: (Feb11/06) Thank you. This is exactly what I meant with my first comment! You illustrate it perfectly. Also note that the Ok/Cancel concept is absent from your pictures which is a good thing. When a parameter is changed it is changed. No need to Ok changes. - job
    • NOTE: (Feb12/06) For the record- this is also exactly what I was talking about. The interface in your images is clean, efficient and uncluttered, and leaves much more usable display area. - Helix
    • NOTE: (Jul29/06) GS: Please let the users design the GUI for you!!. - jedi98

  • MINOR: (Jul3/06) Web UI Caching: Whenever I go the configuration web interface the data in the forms are always stale and I have to force a reload to get the current data. This is reproducable in firefox and IE using an ISA 2004 proxy cache. Recommend using the Cache-Control: no-store header instead of the pragma to fix this problem. -peter

  • MINOR: (Jul20/06) Download SCR XML fails after downloading it about 3 times in a row. This can be fixed with a reboot -ninthclowd

  • TWEAK: (Aug14/06) It's really nice to have SRTP now - however, with SDES the master password is sent unencrypted in the SIP stream AFAIK, so for real security it might be useless... Therefore the GXP-2000 should also support SIPS to make this solid (or is it already supported?) - Mirak

  • TWEAK (Jun05/06) built in microphone When you use the handset and press the speaker button the microphone should not switch to the build in microphone but should stay in the handset. This function is used for letting others listen to a conversation without notice, when the handset is on hook the built in microphone should be used. Right now its not possible to let someone listen to a conversation without lower quality for the called party, even old ISDN phones have this feature.- Datu

    • NOTE: (Jun05/06) Absolutly not! I've never seen a phone behave this way and it would drive me nuts. - nezer
    • NOTE: (Jun06/06) Agree w/ nezer... this is a very odd feature. Perhaps it's common in Europe? I live in USA and i have NEVER seen any phone act this way (and I've seen alot). I suggest a better way of listening is with asterisk and ChanSpy() to listen from another phone. Either way, if this is added it should be optional and disabled by default (If I found a phone that acted in this manner i would report it as a bug.). - Helix
    • NOTE: (Jun06/06) if you want hands free speaking you just let the handset on hook, this is one of the biggest complains with my users, when you want your office mate to listen to a support hotline or client while talking this is very useful. - datu
    • NOTE: (Jun06/06) I Agree. In Europe speakerphones work like this: If you press the "speaker" button and keep the handset off hook, you talk through the handset and you listen both through the handset and the speaker; if you put the handset on hook, you talk and listen only through the speakerphone. And I think this is a nice feature that should be implemented.. - Kurgan
    • NOTE: (Jun06/06) Agree w/ nezer as well... And I do believe this should be listed as a TWEAK and not a MINOR bug as this is not a bug at all since the phone is operating as intended. - ninthclowd
    • NOTE: (Jun16/06) Agree w/ ninthclowd. There's no bug here. It's a difference of oppinion on design. This should be a tweak and the logic should be configurable if possible. I have always found speakerphone logic to be very variable between phones, European or not. - jedi98
    • NOTE: (Jun22/06)I'm calling it. Changed this to a feature request since it seems to me the current behavior doesn't need to be tweaked, but the option to change it should be added as a new feature. Feel free to change if you think otherwise. -NateBell
    • NOTE: (Jul26/06)I cant really live with this bug, and its a bug, when you try to use the hands free mode its not useable, its very low sound, the other party cant almost hear you , its sound like far away, i could rather lay the handset to the desk hand have same results as this hands free mode. Anyway if i hold the handset in my hand, I want the mic in it to be used of course, please correct this bug Grandstream! If any of you think the internal mic should be used then just leave the handset on hook and use it. - Datu
    • NOTE: (Jul26/06) Just because something is listed as a bug or not does not necessarily mean it will or will not get fixed. Classification of a bug is only to let Grandstream know that their product is not working AS THEY INTENDED. The speakerphone mic and handset are working as they designed it (minus the gain issues which ARE bugs and the reason the handsfree mode is so ineffective). Thus what you are asking for is a tweak to change the product not a bug. I personally agree that this could be useful feature and many other people may have use for it, and it would also provide a bandaid for the speakerphone bugs. But if we label everything as bugs instead of tweaks, then grandstream will either A) start ignoring our classification, or B) implement new features without fixing their bugs with the product first. I don't want to start a flame war, so I won't change the classification. I personally think this WIKI should have two classifications: Type (i.e. Bug or Feature Request) and Severity (which could indicate how much the feature is wanted or how severe the bug is). Just stating my opinion =) - ninthclowd
    • NOTE: (Aug04/06) Again, this is a tweak, not a bug. Not even a minor one. - nezer

  • TWEAK: (Jul26/06) Instant Messages Should appear on the screen directly, line by line , so they can send be seen instantly without navigating through a menu. We want to send messages like 'calls forwarded to ...' or 'ringgroup' and see this immediately as instant messaging works like that- Datu


  • TWEAK: (Jul24/06) Getting Parked Calls to work right Works almost perfectly - read on. It could be downgraded to NO TWEAK or eliminated but the notes might be useful to someone. - Anthony
    • NOTE: (Jul20/06) I can confirm that pressing a "Parking Presence" button does not issue a **XXX command, whereas "Speed Dial" and "BLF" do. I have downgraded this note to TWEAK until I better understand the interaction of the Asterisk code, the metermaid patch and the GXP 2000 firmware. - Anthony
    • NOTE: (Jul19/06) The current firmware allows a BLF presence (steady lamp not blinking) to show a parked call but no more. My users want to see a flashing "Parked Call Presence" light/button, press it and retrieve the call. It then resembles a hold button for all extensions. This is what I think many wired current systems have. The current Asterisk method of throwing calls into a parking lot is not going down at all with my users. They refuse to accept this notion. While this is not just a GXP-2000 issue, it is mostly an Asterisk issue - I think the Grandstream firmware should let you pick up the handset, and press a "flashing light" button to pick up a parked call. That means the button/light should not just be a pretty face but be an active button. See my comments above for Asterisk 1.2.9.1 and trixbox 1.1 - Anthony
    • NOTE: (Jul19/06) When you press a flashing button, the phone makes a cal to ** (star star) followed by the extension you're monitoring. Just add dialplan logic to handle that. - julianjm
    • NOTE: (Jul19/06) Yes, I am aware of that. However, I will go back and check it. When you lift the handset and press the button, I dont see any interaction with Asterisk so I am assuming nothing is sent. Furthermore, its not a blinking light when a parking call is present and when the call is finished the light stays on. - Anthony
    • NOTE: (Jul20/06) Its not a blinking light because "show hints" on the CLI says the parked line 71 is "unavailable". Perhaps the metermaid patch did not make it into the trixbox 1.1 install of Asterisk 1.2.9.1 that I am using. - Anthony
    • NOTE: (Jul24/06) I switched to installing CENTOS 4.3 and Asterisk 1.2.9.1 with the metermaid patch from Asterisk 1.2.7.1 which compiles fine on 1.2.9.1. I used firmware 1.1.1.7 for GXP-2000. Set parkedhints=yes after installing the metermaid patch. Program the GXP for BLF on LINE 5 for extension 701, restart Asterisk with /usr/sbin/amportal start, and reboot GXP to register, and you get a steady light because Asterisk "show hints" says 701 is UNAVAILABLE. After the first parked call this light turns off since "show hints" correctly shows IDLE. Asterisk can then be set to understand the BLF button **701. Call parking works as (I) expected. - Anthony

  • TWEAK: (Jul20/06) Download SCR XML should download automatically on reboot if the option is configured, similar to the phonebook xml - ninthclowd


  • TWEAK: (APR02/06) Missed Calls Notification should go away after reviewing them, without having to clear the list. I would like to be able to clear the notification, and then revisit the missed calls list at later points in time to call the people back without having to write the phone numbers down. - Mike
    • NOTE: (JUL13/06) I Agree 100% Mike. Our system uses a ring all queue so missed calls pile on throughout the day. I personally wouldn't complain if there was a way to disable caller ID all together myself. - Mr Esteban
      • Comment: (JUL18/06) Mr Esteban, you can disable missed call notification under the Account information in the web GUI if you want to. I had the same problem, you get 25 calls per hour in a ring group and eventually you'll crash the phones that don't get answered. - ninthclowd

  • TWEAK: (May18/06) Blocked BLF Button. When a BLF is lit(remote user is on the phone) the phone silently ignores when the user presses the corresponding button. What If the user wants to leave a voicemail message for this person or maybe show up as "Call waiting"? Another example is that I use app_devstate from the bristuff package and I use it with my "global DND" function for all my phones. This block also denys me the ability to turn that DND function on and off with the same button. If there must be such a block it should be possible to disable it in the configuration. - Falle

  • TWEAK: (Apr04/06) Dialing from Call Lists. Once you have highlighted a number in one of the call history lists, it would be nice to be able to simply hit SEND to call it, rather than having to hit the little round button to go into another menu, press down until you highlight Dial, and then press the little round button again. On the same note, it would be nice to be able to highlight a number and press DEL to remove it. - thetatag

  • TWEAK: (May10/06) You still can't scroll to the end of a list by hitting the up button, for example on the menu, you have to scroll all the way to the bottom, it would be easier to just hit up to go to the bottom. - mike240se

  • TWEAK: (May14/06) "Do not Disturb" Mode. I'd like to have a better visual feedback for the Do-Not-Disturb-Mode. Now, the DND icon blinks in the display - but if the display backlight turns off, this is almost invisible, especially when not sittig directly in front of the phone. Ideally (in my opinion), the Message Idication LED should flash slowly (very slowly, maybe one short flash every two seconds, to distinguish it from the 'you have voicemail'-flashing!) if the phone is on DND mode. That way, I wont leave the room with the phone on DND, coming back later and forgetting to deactivate DND mode any more ;) - Phloogzoyk

  • TWEAK: (May23/06) Backlight Timeout. I for one would like to see a backlight timeout feature, so that the screen would stop illuminating after 3 minutes of inactivity if you missed a call or something. It would also be nice if the menu system had the same feature. So that if a user accidentally hit the menu button, the menu would exit after 3 minutes of inactivity. Just to preserve screen lifetime. - ninthclowd
    • NOTE: (Jul17/06) I use one in the bedroom and several others through the house. Having the backlight stay ON for a missed call is actually nice as I can see that we need to check the system. There is now a timer to extinguish the backlight after 4-5 seconds of hanging up a call (or clearing the missed calls list). Keep that one in the code - its really nice. Being able to set the timer in the configuration would be even nicer :) - mavila
    • NOTE: (Jul18/06) That may be the case in a low call volume environment, however I have 40+ users and some of them go on vacation or miss a call right after they leave, leaving the backlight for hours at a time. IMHO I believe there should be option to turn off the backlight via timer, just to protect larger investments - ninthclowd









Firmware Notes (Stable 1.1.0.16):


Currently available from Grandstream's Beta Site. This is a browsable directory, more stuff available at Grandstream Beta Site Home.

NOTE: 1.1.0.16 has been moved from the BETATEST directory to a full release version in DOWNLOAD/FIRMWARE ... the betatest directory now contains the preview version. numbered 1.1.1.7, not 1.2.*.* as was suggested earlier.
NOTE: The Users Manual at http://www.grandstream.com/user_manuals/GXP2000.pdf has been updated for 1.1.0.16 as well. -kam

Changelog
Build 1.1.0.16

Fixed IP Fragmentation bug
Fixed GXP-2000/BT500 ring tone change via keypad menu not effective after reboot

Improved audio quality with some audio parameter changes
Fixed we crash on attended transfer on platforms that use To/From headers without square brackets
Fixed we reject cfg files smaller than 512 bytes
Fixed BT-200 keypad UI for TFTP server not working
Fixed a typo in LCP (PPPoE related)
Fixed GXP2000 provides RING only for first incoming call; when first caller hangs up, the ringing stops.
Fixed BT-200/500 does not play short beep on auto-answer
Fixed Bug in Via header when DNS name is used instead of IP address
Fixed we still responds "recvonly" on un-hold SDP message

Changelog listed above taken from Grandstream 1.1.0.16 Release Notes PDF -Helix


Bugs / Tweaks (1.1.0.16):


  • MAJOR(Aug02/06) Bug in IP implementation: The firmware incorrectly treats certain IP address as a local broadcast address which prevents it from communicating with devices with such IP addresses. It looks like the firmware does this: if dest_ip & ~netmask) == ~netmask) it_IS_broadcast; instead of this: if (dest_ip == (my_ip it_IS_broadcast; For example, if your IP address is 1.2.3.162, netmask 255.255.255.224 and gateway 1.2.3.190, a packet to 4.5.6.31 should go through the gateway, but the phone considers it a local broadcast and sends it to ff:ff:ff:ff:ff:ff ethernet address. The longer your netmask is (i.e. the smaller your subnet is), the more destinations you will be unable to reach. I have reported this bug to support@grandstream.com when 1.1.0.13 was the latest stable and again when 1.1.0.16 came out, but I have not received any reply (except for the "ticket generation") from them. Jaroslav Janacek

FIXED 6/29/06 The low volume on the speakerphone when paging is fixed. The new volume level works fine for me. Anthony ___

  • MAJOR: (Jun19/06) Mute after putting a line on hold: If you are on a call and you put the party on hold, then you recieve a second call on another line but don't answer, when you go to take line 1 off hold they will not be able to hear you. Please note that I built a rollover macro(in asterisk) so you can use one SIP account for all lines on the Grandstream and new calls will come into any available line even while your on the phone. It's possible this could be a software setting but I believe this to be hardware, especially because of the workaround listed below. - ninthclowd
    • WORKAROUND: Tap the handset release button and they will be able to hear you again. - ninthclowd
    • NOTE: Why do you need a macro to receive multiple calls on a single account? GXP-2000 can handle up to 11 calls on a single account without any special setup. Or maybe I have not understood what you are doing. - Kurgan
    • NOTE: (Jul03/06) It's so Asterisk will still dial the phone even though it's reporting BUSY. I believe this is not really relavent in retrospec - ninthclowd

  • MAJOR: (Jul01/06) Very soft outbound volume through handset: Apart from the problems with the speakerphone, the volume of the caller to the callee is extremely soft when talking via the handset. I test by calling myself via GXP2000 -> Asterisk (SIP - g729 or alaw) -> VSP (alaw) -> PSTN and placing my PSTN phone on mute while speaking into GXP2000. I have tried various providers and the voice level is the same low output using GXP2000. When I make the same test calls via SPA3K the volume is perfectly loud and clear. I don't even find the speaker volume on GXP2K increases the mic level. I found this when doing testing with firmware 1.1.0.13 and find it continues with this version. I hope individual RXGAIN/TXGAIN is on the way! - Giulio
    • NOTE: (Jul03/06) I agree. This is a must fix. - Anthony

  • TWEAK: (May10/06) You still can't dial a number when the phone is on hook by dialing the number then hitting send or speakerphone. - mike240se
    • NOTE: (Jul03/06) On hook dialing has been open for some time in the feature request section, please post any comments on this topic there as to keep everyone on the same page. - ninthclowd

  • MINOR: (Jul5/06) UI: When picking up line appearances 10 or 11 the display shows LINE:: and LINE;: instead of LINE10: and LINE11:. - Andrew

  • MINOR: (Jul18/06) BLF: When you have an external number programmed into speed dial key 1 configured as a normal button and then have subsequent buttons configured for BLF the phone tried to register the number associated with key 1 as BLF also. First noticed with 1.1.0.13. Currently have a ticket open with grandstream. - Gareth

  • TWEAK: (Jun30/06) Add support for SIP header Alert-Info to trigger custom ringtones, e.g. SIPAddHeader("Alert-Info: ring3"). This was a feature request and can be seen further below on this web page. I decided to stick my neck out and bump it up since the absence of this feature bugs me. Feel free to shoot me down. I would like to see an easy way to distinguish internal vs external calls via ringtones on the same extension line. It seems to me the best answer is SIPAddHeader - are there other easy solutions? - Anthony
    • NOTE: I strongly agree! This is a feature I am been waiting for since when I bought my first GXP-2000. Many of my customers want this! - Kurgan
    • NOTE: This may already be possible (Jul 01/06) According to a Grandstream FAQ, this is already possible. Set the 'distinctive ring number' for rings 1-3 to a word (soemthing thats not a number), then send that word as alert-info. I will try this if i get a chance later, but it would be very cool if it were true! -Helix
    • NOTE: It works! Use something like this: exten => 12345,2,SetVar(ALERT_INFO='<ignored text>\;info=ring_word'). Then you have to put the "ring_word" as the distinctive ring number in the phone's config. It would be even better if the phone could simply recognize "ring1", "ring2" and "ring3" as Alert-Info values, but it does not. You have to set up distinctive ring strings in phone's config to make it work. - Kurgan
    • NOTE: It works! The above did not work for me, but the following did: exten => 12345,2,SetVar(_ALERT_INFO='<ignored text>\;info=ring_word'). I use A@H 2.5 which has Asterisk 1.2.0 .. - Anthony
    • NOTE: (Jul03/06) This is not a bug. Relabeled as a tweak. - ninthclowd
    • NOTE: (Jul04/06) Setting variables depends on Asterisk version, I forgot to mention that I use Asterisk 1.0.7. - Kurgan
    • BUG? (Jul04/06) It seems that only ringtone 2 works using alert_info. If I try to use ringtone 3 I get the default ringtone (1). So I end up with TWO ring tones (default and ringtone 2) and not three. Could someone check and report back? - Kurgan
    • BUG? (Jul05/06) I agree. For me works only custom ringtone 1. So only 2 ring tones are available (even better than 1 :-) ) - maxx

  • TWEAK: (Feb24/06) Speakerphone volume setting should be separate from handset volume. I need to turn the speakerphone up to full volume to hear it, but this means the handset is too loud. - bani

    • NOTE: (Sept04/06)
If you have two numbers registered on the phone can one be made to glow green and one made to glow red depending which extension is ringing. The reason for asking, If you have two company names running on one phone, then when more than two calls dome in you cannot distinguish which line is for which extension as the last call always shows on the display and all lights are red.









Firmware Notes (Beta 1.1.0.13):


Currently available from Grandstream's Beta Site
  • NOTE: (May18/06) The above link seems to be broken. Anybody know where to get it? - ninthclowd
  • NOTE: (May18/06) Link fixed. Grandstream changed one of the folders on their site. The new folder (/BETATEST/GXP2000_BT200) seems to suggest perhaps an upcoming product? Another file in the folder has firmware for bt200 and bt500 products also... can't wait to see them! - Helix
  • NOTE: (May19/06) I didn't happen to see the release notes bundled in this package. Anyone have a copy they can post? - ninthclowd
  • NOTE: (May19/06) The site is a browsable directory, click here and you can find all sorts of cool stuff. The files you want are in the GXP2000_BT200 folder. -Helix
    • NOTE: (May22/06) Thanks for the link! -ninthclowd

Changelog
Build 1.1.0.13 5/16/2005
  • Add Quick IP Calling mode
  • Fixed GXP-2000 Speed Dial/Asterisk BLF pick up broken in 1.1.0.12

Build 1.1.0.12 5/15/2006
  • Fixed GXP-2000 crashes when a very long DTMF string is dialed
  • Fixed SIP NOTIFY to event REFER violating RFC 3515
  • Fixed we do not affix To-tag for PRACK request
  • Fixed we do not use new branch for PRACK request
  • Fixed we do not include Contact header in 180
  • Fixed we do use random port for RTP even if random port is set to yes

Changelog listed above taken from Grandstream 1.1.0.13 Release Notes PDF included with 1.1.0.13 firmware -benny23


Bugs / Tweaks (1.1.0.13):

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!
(In addition to adding user feedback here, please copy items from the previous bug/tweak list to this list if you discover that they are still issues and note that you have done so! Thank you!)

  • MAJOR: (Jun19/06) Mute after putting a line on hold: If you are on a call and you put the party on hold, then you recieve a second call on another line but don't answer, when you go to take line 1 off hold they will not be able to hear you. Please note that I built a rollover macro(in asterisk) so you can use one SIP account for all lines on the Grandstream and new calls will come into any available line even while your on the phone. It's possible this could be a software setting but I believe this to be hardware, especially because of the workaround listed below. - ninthclowd
    • Workaround: Tap the handset release button and they will be able to hear you again. - ninthclowd

  • MAJOR: (Jun13/06) Speaker phone does not work at all and hangs when doing a soft reboot (from the web server). MAC is 00.0B.82.09.75.13 - Drew
  • MAJOR: (Jun09/06) When using the GSM codec, silence supression is always on, regardless of the setting in the options. - Adrian2k
  • MAJOR: (May22/06) FWIW, Screen still blanks! MAC is 00:0B:82:03:A1:25 - MattB
    • NOTE: (May22/06) In the 40 phones that my company purchased, one of them seems to blank. I'm wondering if maybe this is the result of a partially defective unit rather than firmware issues. - ninthclowd
    • NOTE: (May22/06) Once again my screen blanked. So far it has only happened once after 3 days. Look like the blanking is still an issue, but not as bad as it used to be. -NateBell
    • NOTE: (May25/06) This might help diagnose - my screen was blank again this morning. A call came in for me (whilst the screen was blank) and I hit MUTE/DEL to send to Voicemail. The second I pressed it the screen came back again! - MattB
    • NOTE: (May25/06) Screen blank workaround Further to my previous entry, whenever your screen blanks just press the MUTE button - brings the screen back every time. - MattB
    • NOTE: (Jun23/06) One of my phones with a MAC of 00:0B:82:03 blanks immediately after the Grandstream logo flashes by on bootup and nothing seems to bring the screen back after this happens. The screen works fine on the older 1.1.0.1 firmware (perhaps the 1.1.0.11 as well, but I haven't verified that) - Flu
    • NOTE: (Jun29/06) Phone with a MAC of 00.0B.82.03.D4.32 blanks, generally triggered by an incoming call. But not every call, this is intermittant. MUTE/DEL button does not seem to bring the screen back, only a reset will do it. The last version that worked (mostly) was 1.0.2.13 - Jedi98

  • MAJOR: (May22/06) Quiet Speakerphone. I might get flamed for this, but I am elevating this to a major bug. The speakerphone continues to be esentially useless. Yes, the echo problem it had under 1.0.1.9 has gone away, but it is too quiet to be usable in all but the most silent environment. If you have any ambient noise, the speakerphone experience is ruined. And even in silence, it is very quiet - even at the loudest settings. The speaker is certainly capable of producing much louder sound, as is evidenced by the maximum ring volume, and a good echo cancelling algorithm should be able to handle the added volume being picked up by the microphone. In my business deployment, this is the number one complaint about these phones. - thetatag
    • NOTE:(Jun14/06) I agree with this. I could get by with paging a phone for which I could set the volume. Unfortunately, a soft or hard phone reboot will reset the volume to default and render the page inaudible. I would like to see a programmable volume control which might be a first step towards a fix. - Anthony
    • NOTE: (Mar23/06) I'm going to have to agree. The single biggest complaint I recieve about these phones from my end users is about the speakerphone issues. It is in fact quiet, however i think the bigger problem is the lack of clarity. It seems like the speakerphone quality just gets worse and worse with each release (but at least there isn't any echo anymore) - ninthclowd
    • NOTE: (Jun21/06) I hate to backtrack but now after playing with it more, I believe the quality issues are directly related to the speakerphone being so quiet. Therefore I would like to state that the bigger problem is the volume. - ninthclowd

  • MINOR: (May22/06) Call Distortion: Since I updated with this firmware all phone calls sound distorted on my end, but the other party hears me fine. Reverting back to the previous firmware did not solve the issue, which leads me to believe that it may not be possible to revert back once this firmware is loaded. - Mike B.
    • NOTE: (May29/06)I also have this problem with the current firmware. -Richard H.

  • MINOR: State Bugs. Loopback call (dial phone from itself, via asterisk) drops out on answer (as expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up arrow adjusts the volume instead of missed call list. -Jedi98
    • NOTE: (Mar21/06) Confirmed the problem on the latest beta firmware "1.0.2.13". -ChrisUK
    • NOTE: (May22/06) Reproduced on 1.1.0.13. - job

  • MINOR: (Mar29/06) Numbers off-screen. When entering long phone numbers (i.e. for international calls) the phone number goes off-screen as it advances, rather than moving up to the upper line as it used to. - Mike

  • MINOR: (Mar29/06) Ring Volume. Ring volume still does not survive reboot - it resets back to middle setting. - Mike

  • MINOR: (Mar31/06) Muted DTMF. You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag
    • NOTE: (May24/06) Reproduced with 1.1.0.13. - vgster



Firmware Notes (Beta 1.1.0.11):


The firmware 1.1.0.11 is attached to this page:
Image


Changelog
Build 1.1.0.11 4/28/2006
  • Fixed the ping problem when the device is in router mode
  • Enabled the broadcast drop mode (this should improve the switch performance for multicast and broadcast)
  • Fixed the 3-way conferencing issue when the re-invite to bring the 1st party out of hold status gets challenged.
    • This is difficult to verify. Basically our 3-way conferencing can be broken into the following steps:
      • A invites B
      • A re-invites B to put B on hold
      • A invites C
      • A re-invites B to put B out of hold status
    • If at step d) the proxy challenges the request with 401/407, we couldn’t complete the conferencing
  • Fixed the PPPoE TCP problem
  • Added idle timers to fix more idle screen blackout cases
  • Fixes for blank LCD or corrupted GUI issues
  • A-Tick and DC filter changes
  • Fixed crash on incoming call when all channels are in use
  • Fixed lost registration problem
  • Fixed we will never switch DNS server even if primary DNS server failed to respond and there is a secondary DNS server
  • Changed for GXP-2000: once entering direct IP calling mode, the cursor focus is in the text field instead of CANCEL button
  • Reduce the GXP2000 handset earpiece audio level by 4.5 dB
  • Fixed GXP-2000 crashes when using MISSED CALL GUI to dial out
  • Added use of MUTE/DEL key during incoming call ringing state will reject call using 486
  • Added MUTE/DEL key will act as toggle key to turn DND on and off during idle
  • Fixed GXP-2000 direct IP call cannot specify port.
    • Note that a new input method is specified here: you will use * to enter dots (separator between octets) and use # to enter colon (separator for port). So you can enter "10.10.12.135:5068" using "10*10*12*135#5068". This is probably more intuitive
    • Note 2: Direct-IP calling feature is further cleaned out so that STUN mapped info is not used when we detect direct IP calling destination is in local subnet (From and Contact headers are also cleaned to use IP address only, not including the configured SIP URI).
  • Fixed AGC setup change
  • Fixed RTPSend bug
  • Fixed we do not use the previous SSRC, timestamp, and sequence number after restoring a previously hold call
  • Fixed static IP problem in 1.1.0.2
  • Fixed we start sending RTP when restoring a call before we receive 200 OK Fixed we do not clear out CallFwd settings when user configure to disable call features
  • Added special factory workaround mode-when configured to use static IP 192.168.0.160 and no gateway IP is configured, provisioning is skipped
  • Added support for Broadsoft Click-to-Answer feature using "talk" event
  • Fixed GXP-2000 cannot make direct IP calls
  • Fixed GXP-2000 factory MAC-Edit function cannot change last digit to A-F
  • Fixed redial does not append the dial-plan prefix
  • Fixed we will retry 5 times if only config server is configured and there is config file
  • More LCD fix for GXP-2000
  • Fixed some crashes Issues

Changelog listed above taken from Grandstream 1.1.0.11 Release Notes PDF included with 1.1.0.11 firmware - flu



Bugs / Tweaks (1.1.0.11):

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!
(In addition to adding user feedback here, please copy items from the previous bug/tweak list to this list if you discover that they are still issues! Thank you!)


  • MAJOR: (APR17/06) Screen still blanks! MAC is 00:0B:82:03:A1:25 - MattB
    • NOTE:(May05/06) Updated: I can confirm my phone's screen blanked this morning. My MAC address is 00:0B:82:03:A0:??. -Nate Bell
    • NOTE:(May06/06) My phone does *not* have blanking problems. Mac 00:0B:82:05:AA:xx - Kurgan
    • NOTE:(May08/06) blanlking is now mostly corrected .. went blank during a call a few times, but mostly is now OK. Previously was just blank all the time or upside down etc, so a BIG improvement. - Robin Sz
    • NOTE:(May08/06) Just to clairfy, with 1.0.2.13 my screen would blank after a couple seconds upon powering on. I reverted back to 1.0.2.8 where my screen would blank after a day or so. I still have blanking issues with 1.1.0.11, but so far it has only happened once, and I've been running the firmware since it was released. So I'm still having a problem, but as Robin said, it has been much improved for me. -NateBell
    • NOTE:(May12/06) (possible hardware issue) Since 1.0.1.11 my old gxp (mac 00.0B.82.03.CD.xx) LOCKS up... a LOT. Used to blank once every few days, now it locks up (screen is frozen but image still present, clock stopped) several times per day, often within a few minutes of powering it on. Tried factory defaults, flash back to 1.1.0.1 w/ no luck. I think my phone may just be dying, but figured I'd toss this in anyway in case it helps. -Helix
    • NOTE:(May12/06) I can confirm my phone's screen blanked too. My MAC address is 00:0B:82:03:CC:2E. -cervajs
    • NOTE:(May17/06) Blank screen too. My MAC: 000B8203CC2F. -Festr

  • MAJOR: (May 3/06) Still problems with BLF in Firmware 1.1.0.11. No lights blink or lighten up when other partner is calling or getting a call. This was fixed in a very previous version 1.0.2.3. I tested with two GXP2000. I can see in Asterisk CLI (show hints) that the hints are active and working but the Watchers still 0.- FreshSmith
    • NOTE: (May04/06) Do you have an old GXP? My phone blinks and lights up like an x-mastree when other users are on the phone. - Falle
    • NOTE: (May04/06) Seems ok on mine - ninthclowd
    • NOTE: (May05/06) Mine is ok too. What are the MACs on your units? - qortra
    • NOTE: (May05/06) This isn't much help, but mine are working too. -Nate Bell
    • NOTE: (May06/06) Mine works too. Mac 00:0B:82:05:AA:xx, Astersik 1.0.7 (Debian Sarge) - Kurgan
    • NOTE: (May06/06) Sounds more like a config error, did you set subscribeextension= in sip.conf? - mike240se
  • MINOR: (May05/06) Ring Volume still not survive to a reboot, but now, the default volume is lower... - Snipefoo
  • MINOR: (May05/06) Ring Tone needs a reboot when setting from the phone menu, but not when setting from de web UI - Snipefoo
  • MINOR: (Mar31/06) Muted DTMF. You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag
  • TWEAK: (May04/06) Blocked BLF Button. When a BLF is lit(remote user is on the phone) the phone silently ignores when the user presses the corresponding button. What If the user wants to leave a voicemail message for this person or maybe show up as "Call waiting"? Another example is that I use app_devstate from the bristuff package and I use it with my "global DND" function for all my phones. This block also denys me the ability to turn that DND function on and off with the same button. If there must be such a block it should be possible to disable it in the configuration. - Falle
  • MAJOR: (May05/06) Call mute when switching lines. When using multiple lines for one extension (i.e. for rollover purposes) the phone will sometime mute one or both of the parties when switching lines. At least I assume it's a mute. This was not an issue in 1.0.2.13 and previous - ninthclowd
    • NOTE:(May06/06) Could you please explain in a more precise way what do you do to produce the bug? - Kurgan
    • NOTE:(May09/06) I'm not exactly sure what's causing it, but I&#039;ll tell you how I'm setup. I set up line 1 to have a SIP account. Line 2,3 and 4 I leave default. Then when I'm on a call on line one, I can dial out on line 2 but the remote party will sometimes get muted. They will be able to hear me but I can't hear them. The call continues and the timer still counts and I can switch to and from line 1. I find that it seems to happen when using the line buttons to switch between lines. - ninthclowd
    • NOTE:(May09/06) So we're talking loss of one/other/both RTP stream maybe. One idea, and it's a real stab in the dark, is the codec being re-negotiated? If any of it goes via Asterisk you can probably get verbose messages from there. What about the syslog? - Jedi98
    • NOTE:(May10/06) What should I be looking for in the log? I did find some of these "NOTICE[6998] rtp.c: Unknown RTP codec 100 received" don't know if thats normal or not but I don't remember seeing them before. - ninthclowd
  • FEATURE: (May14/06) I'd like to have a better visual feedback for the Do-Not-Disturb-Mode. Now, the DND icon blinks in the display - but if the display backlight turns off, this is almost invisible, especially when not sittig directly in front of the phone. Ideally (in my opinion), the Message Idication LED should flash slowly (very slowly, maybe one short flash every two seconds, to distinguish it from the 'you have voicemail'-flashing!) if the phone is on DND mode. That way, I wont leave the room with the phone on DND, coming back later and forgetting to deactivate DND mode any more ;) - Phloogzoyk





Firmware Notes (Alpha 1.1.0.4):


This is an internal build correcting issues in the current Beta. Distribution of this build has been prohibited by Grandstream. (Expect to see a public release version soon.) - thetatag

Changelog
Build 1.1.0.4 3/24/2006
  • Fixed GXP-2000 crashes when using MISSED CALL GUI to dial out
  • Added use of MUTE/DEL key during incoming call ringing state will reject call using 486
  • Added MUTE/DEL key will act as toggle key to turn DND on and off during idle (These two are the sweetest part most people like)
  • Fixed GXP-2000 direct IP call cannot specify port.
    • Note that a new input method is specified here: you will use * to enter dots (separator between octets) and use # to enter colon (separator for port). So you can enter "10.10.12.135:5068" using "10*10*12*135#5068". This is probably more intuitive.
    • Note 2: Direct-IP calling feature is further cleaned out so that STUN mapped info is not used when we detect direct IP calling destination is in local subnet (From and Contact headers are also cleaned to use IP address only, not including the configured SIP URI).
  • Fixed AGC setup
  • Fixed RTPSend bug
  • Fixed we do not use the previous SSRC, timestamp, and sequence number after restoring a previously hold call

Bugs / Tweaks (1.1.0.4):

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!

  • MINOR: (Mar31/06) Muted DTMF. You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag

  • TWEAK: (Mar31/06) MUTE/DEL Call Refusal. Currently, if you press MUTE/DEL while the phone is ringing, it immediately responds as busy. However, if the phone is ringing while you are on the phone (call waiting scenario), the MUTE/DEL button mutes your current call. Personally, I would like to see it also respond busy even if you are on the phone when you press it. How likely is it that you will want to mute you current conversation while another call is ringing in? It seems more likely that you would want to refuse a call-waiting call without having to hold your current call. This is subjective, though, so I'd like to see a little feedback here before recommending this behavior to Grandstream. - thetatag
    • NOTE: (Mar30/06) How about: press MUTE/DEL quickly to mute the call, hold it down (more than 1 second) to toggle DND status while on a call. - bani
      • NOTE: (Mar31/06) The MUTE/DEL button now has three functions. It toggles DND status (when the phone is idle), it refuses an incoming call (when the phone is on-hook and ringing), and it mutes the current call (when the phone is off-hook and in an answered state). I like your idea, except I would prefer the longer button-press toggle DND when no call-waiting lines are ringing and reject an incoming call if there is a call-waiting line ringing. Although, personally, I only see marginal value in being able to toggle DND status while on the phone. - thetatag

  • TWEAK: (Apr03/06) Daylight Savings Time. Daylight savings time is implemented in the GXP-2000 by simply setting the time an hour later when the option is selected. This results in having to change this setting on every phone (or a global configuration file) and rebooting every phone to update the time. (The time may update without a reboot, but it does not do so immediately.) The GXP-2000 should have daylight savings time implemented like nearly every other smart system: it knows when to turn it on and off based on the date. The option, then, should be "Automatically adjust for daylight savings time," and the phone should take care of changing at the appropriate times automatically. - thetatag
    • AGREED! - Kurgan
    • Ya manually changing DST on 40 something phones was not that fun. - ninthclowd
    • NOTE:(Feb04/06) I am a bit skeptical. If this cannot be done reliably it should not be done at all. Can Grandstream input the local rules for the whole world? How can the phone know Australia postponed DST this year? But if they let you edit the DST rules manually it is a good idea. My computer does this reliably so perhaps it is possible. - job
    • NOTE: (Apr04/06) NTP handles daylight savings time. - bani
    • NOTE: (Apr05/06) This won't apply to all of us, but according to a thread on the AAH forums, you can have the gxp-2000 get time from an AAH server by placing the servers IP into the gxp's NTP setting (NTP is setup and enabled by default on AAH, and likely will work on a straight Asterisk installation if NTP is enabled as well). - jehowe
    • NOTE: (Apr06/06) What do you mean NTP "handles" DST? NTP (or actually SNTP here) synchronizes time. That time is UTC time. How the client interprets that into the displayed time which must be corrected for time zone and DST has nothing to do with the synchronization protocol. - job
    • NOTE: (Apr04/06) Err yes you are correct. Hardcoding rules for every locale on the planet is not practical, so grandstream should just put in options to configure the start and end dates for DST. Even DST for the US will be different in 2007, thanks to the Energy Policy Act of 2005. - bani
    • NOTE: (Apr27/06) You can do this centrally on DHCP (presuming you have admin of your DHCP server) by setting the time offset (option 2) and allowing it to override the TZ on the phone. Also set the phone DST off. Then you only need to change the DHCP time offset on the time change. - jedi98
    • NOTE: (May03/06) Good idea, it has also been implemented in beta 1.1.0.11, but won't this break other clients (not phones) that get timezone from DHCP? I mean, here I am changing timezone, not DST settings. Won't other clients get a wrong timezone and get confused? - Kurgan
    • NOTE: (May08/06) Good question. I tested this, admittedly only with GMT, but it has not caused a problem here. I think this is because time-offset, on DHCP, when used correctly by a client replaces the TZ rather than adding to it. You should be alright as long as all your clients are in the same TZ. However, as always, you should test before applying this to any critical environment. - Jedi98



  • TWEAK: (Apr04/06) Auto Call Waiting. In asterisk there seems to be no way of setting CW by default. Grandstream should add an option to have the phone run a set of commands just after registering...... such as *70 to enable CW (Call Waiting) or *71 to disable, *72 for CF (Call forwarding) or *73 to disable CF - naturalblue
    • NOTE: (Mar31/06) Why would you want Asterisk to handle Call Waiting? This is a multiline phone. I propose to remove this request unless better explained. - job
    • NOTE: (Apr04/06) This tweak request doesn't make any sense (and it's unsigned) so i'm moving it to the bottom. - bani
    • NOTE: (Apr04/06) I think this is a good idea. My GXP2000 has CW enabled and yet it still doesn't accept the next call in unless i enable call waiting in asterisk. As this phone is not just for simple 1 line connections and has been marketed as an enterprise phone i think the ability for it to easily integrate in PBX solutions such as asterisk is very valuable. Also it would allow for the phone to run multiple features in the future by simply running a command after registration. i have signed this post and my original, i thought the wiki would add it for you as a registered user. - naturalblue
    • NOTE: (Apr04/06) You must be using asterisk@home or something — call waiting is not something asterisk "does", and a phone's job should not be to enable/disable features on the PBX. The PBX should do that itself. This is not a feature that belongs in the phone, someone should fix asterisk@home or whatever you are using. - bani
    • NOTE: (Apr05/06) Same as bani said pretty much- for a SIP channel, * will always have call waiting unless it's turned off (possibly with a call limit). When a call is in progress on a SIP channel and another call comes in, * tells the SIP endpoint (ip phone) that another call is coming in (this is considered default behavior for SIP). The GS deals with this by playing a call waiting tone, printing info on the display, and making one of the other LINE lights flash (even if the call is not coming in on that button's account, the line it comes in on is displayed on the LCD). You can then if you want put your current call on hold and switch back and forth between the two calls. If * or your voip provider are not sending the second call, then they are misconfigured. Perhaps you have it set up like a fax line which usually shouldn't get CW? Or if you're using Asterisk@home perhaps it has a funky feature set... either way while dialing something on registration might be useful for some hacks, it's not required to make call waiting work. Hope that helps :) - Helix

  • TWEAK: (Apr04/06) Dialing from Call Lists. Once you have highlighted a number in one of the call history lists, it would be nice to be able to simply hit SEND to call it, rather than having to hit the little round button to go into another menu, press down until you highlight Dial, and then press the little round button again. On the same note, it would be nice to be able to highlight a number and press DEL to remove it. - thetatag

  • TWEAK: (Apr10/06) Some useful things It would be nice if phones would support that:
    • Call forward if no answer after settable timeout
    • Dialing from call list by just picking up handset or pressing speaker or send button(and maybe #)
    • Option to set busy trigger to limit max incoming calls at same time
    • set callforward call by pressing transfer button (instead of dialing *72) and number (onhook - transfer all, off hook - transfer if busy)
    • If down key is pressed, phonebook appears on the beginning. Why wouldn't up key display phone book on the end? Show missed calls could still stay up, because first you check missed calls, right? - bad2Dbone

  • TWEAK: (Apr16/06) DHCP Option 66. This option for allowing dhcp to override tftp should be set to on/yes by default. You can set the provisioning of this (using p145) in the cfg file for the phone but this is no use as the phone wont go to your tftp server to get the cfg file until you manually log into the phone and set this option to yes. If provisioning/setting up lots of phones this causes quite a delay. - naturalblue


(In addition to adding user feedback here, please move items from the 1.0.2.13 bug/tweak list to this list if you discover that they are still issues! Thank you! - thetatag)



Firmware Notes (Alpha 1.1.0.1):


WARNING:
(Mar29/06) Grandstream has stated that this firmware version was not authorized for public release. It was released at a work-in-progress state to a few individuals with particular firmware issues to verify that a change in one piece of code would correct their problem. If you choose to use this version, be warned that there may be known broken functionality, bugs, or unstable developmental code. You have been warned! - thetatag

This firmware rev was posted to the bottom of this page a few days ago and apparently nobody noticed. Unknown if distribution is sanctioned by Grandstream, assuming it's ok because it's been posted here for a few days. - Helix

Also note that you can revert from 1.1.0.1 back to 1.0.2.13 (software) and 1.0.2.3 (bootloader) should this cause problems for you. -Mike

The firmware 1.1.0.1 is attached to this page:
Image


Changelog
This build was supplied without a changelog. Perhaps someone can add it?
Attempts to fix early-model GXP display corruption issue are not succesful.
  • Added: two new options in the Special Feature drop-down of the account page: RNK and Sylantro. -Mike
  • Changed: default display screen now shows name and number instead of date (when time is in top right). (Please correct me if this was already included in 1.0.2.13, however, I never noticed it there.) - Mike


The ZIP also has two additional files- bt200a.bin and bt500a.bin. I don't think my GXP asked for them. Maybe these are related to the new sidecar, or upcoming lower-end phone models?
  • Note: The naming convention suggests that the other two files mentioned above are firmware releases for (upcoming?) BudgetTone phones 200 series and 500 series. -Mike

Bugs / Tweaks (1.1.0.1):

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!

  • MAJOR:(Mar30/06) Conference Call-MUTE. When in conferenced call unable to Mute phone. - MikeB
    • NOTE: (Apr03/06) Some months ago I posted the bug that muting while on a conference call caused one or both of the remote parties to not be able to hear the other (as opposed to just muting the microphone on the local phone). I wonder if rather than fixing the problem, they just decided to eliminate mute functionality on conference calls. If that's the case, it is certainly the wrong answer. - thetatag

  • MAJOR: (Apr11/06) Blanking Screen. Phone's LCD becomes blank after a few hours or so on a 00.0B.82.03.xx.xx MAC addresses phone, it seems to happen alot less (used to happen 3-4 times a day now its more like once every 1 or 2 days) than with 1.0.2.3 firmware but it is still happening, my phone rings for every phone call coming into our company (i dont answer it just rings and reception pick it up,) the other phone we are using doesnt do this and i havent yet seen that phone blank with this firmware... in other words the blanking problem is directly related to the number of calls sent to the phone (this was also the case with 1.0.2.13) . - SoloFlyer
    • NOTE: (Apr10/06) Just wanted to mention that I have not been experiencing this with mine on a 00.0B.82.04.xx.xx MAC phone. - Mike
    • NOTE: (Apr16/06) Worse than ever on my 00.0B series mac phone ... Display lasts about 3 seconds after reboot before going blank, regardless of whether plugged intot he network or not ... a backwards step again I'm afraid. How I wish I had never upgraded :( . - robinsz
    • NOTE: (Apr18/06) I just came back from a 4 day weekend and the phone hasnt blanked, but since there wouldnt have been receiving phone calls, i belive it confirms my theory that its related to phone use. - SoloFlyer
    • NOTE: (Apr21/06) My experiences with this firmware support the going theory. I have 2 phones running 1.1.0.1, one that gets a high volume of calls (10-20 per day) and one that gets almost no calls. The called phone usually has a blank screen in the morning, while the other phone doesn't. -Nate Bell

  • MINOR: (Mar29/06) 503 error. If the phone receives SIP/2.0 503 Service Unavailable it will print 503 in the display. It should print Unavailable instead. How to reproduce: make exten => 5555,1,Congestion in asterisk extensions.conf, then dial 5555 from the gxp2000. - bani

  • MINOR: (Mar29/06) NTP Broke NTP doesnt work at all and phone seems stuck on January 2nd at 12am everytime i look at it. Since there is no way to change the date/time manually, you just have to deal with no date and time. - mike240se
    • This may be an issue with your phone or the individual NTP you are trying to connect to. My NTP is working fine (try time.nist.gov). - Mike
    • I tried setting it to time.nist.gov and also to an IP, I also tried assigning it via DHCP (and changed the option to allow it in the web gui) and no go, stuck on january 2nd. With 1.0.2.13 all the phones work great with NTP and with 1.1.0.1 all the phones don't work so I dont think its an actual phone problem or a network problem since reverting to 1.0.2.13 have the problem. Also if you read in the comments at the bottom of the page, other people have the same problem. Maybe only newer phones are affected? Mine are all pretty new. - mike240se
    • NOTE: (Mar29/06) Make sure you don't have DHCP option 42 overriding the phone's NTP settings. - bani
    • NOTE: (Mar30/06) I triple checked and tried everything, it just keeps going back to January 2nd after every reboot, I downgrade the firmware and NTP works perfect. - mike240se
    • => Workaround: (APR02/06) I discovered what the real bug is here (I think) :). I'm getting the same thing after changing the daylight savings time option to yes. What I needed to do was select Daylight Savings to no, and change my timezone instead (to fake the right time). It has to be a bug in the Daylight Savings option. - Mike
    • NOTE: (Apr02/06) I can't reproduce, I've selected my time zone and daylight savings works as expected. - job
    • NOTE: (Apr07/06) I also can't reproduce this NTP problem, and setting daylight savings settings makes no difference to me... - SoloFlyer
    • NOTE: (Apr07/06) You guys fixed... err, jinxed me: now I can't reproduce it anymore either. Maybe it was a momentary glitch with my network or phone connecting to the NTP server. Anywho, I'm glad its working. Should we mark this as a non-issue now, or is anyone still experiencing this problem? - Mike

  • MINOR: (APR17/06) When turning on the account option: "Unregister on reboot" the phone reregisters every 30 seconds regardless of timeout settings. With 1.0.2.8 this was also pressent but the reregistrations occured every 60 seconds. When "Unregister on reboot" is set to no the phone works like it should. - Falle

  • TWEAK: (APR05/06) Time Display while on Call: I would like to see the time displayed while on a call in the upper right corner of the display, regardless of off-call preference. More often than not I'll find myself making appointments or what-not and will be looking for the time (and not the date, since I have it set to show time in large when not on a call. - Mike

  • TWEAK: (APR17/06) When a BLF is lit(remote user is on the phone) the phone silently ignores when the user presses the corresponding button. What If the user wants to leave a voicemail message for this person or maybe show up as "Call waiting"? If there must be such a block it should be possible to disable it in the configuration. - Falle



Firmware Notes (Beta 1.0.2.13):


This is an internal build correcting issues in the current Beta. Distribution of this build on this Wiki has been sanctioned by Grandstream. - thetatag

The firmware 1.0.2.13 is attached to this page:
Image


Changelog
Build 1.0.2.13 2/21/2006
  • Fixed LCD problem on older units
  • Fixed keypad not responding if boot up without network cable
  • Fixed when LINE keys are pressed when in GUI MENU, the usual "Dial Using" prompt is not displayed
  • Added support for DHCP option 2 (Time Offset). Provision variable P143, possible values 0/1. When set to 1, it will override the configured Time Zone setting if available. Default is No (0).
  • Added support for DHCP option 42 (NTP server). Provision variable P144, possible values 0/1. When set to 1, it will override the configured NTP server. Default is No (0).
  • Added support for DHCP option 66 (TFTP server). Provision variable P145, possible values 0/1. When set to 1, it will override the configured provision path and method. Default is No (0).
  • Added configurable DHCP options 12 (host name/P146), 15 (domain name/P147), and 60 (Vendor Class ID/P148). Max length allowed is 32 bytes each.
  • Fixed we do not follow the port in the Record-Route URI if there is maddr attribute
  • Added we will attempt to start the initial provisioning routine every minute for 5 times unless we received any response from the server (any type of HTTP error or OK response, and TFTP error or data response will stop further trying)
  • Changed we will restart STUN accounts when STUN port mapping changed (previously we only check IP address change). Also changed: STUN checking interval is now the same as the configured keep-alive interval (used to be 1 minute fixed)
  • Changed the RTP keep-alive interval (when a call is placed on hold) to the same as the configured keep-alive interval (used to be 25 seconds fixed)
  • Fixed we do not send RTP keep-alive interval during MUTE
  • Fixed when early-dial is in use, the phone times out 4 seconds after the last incomplete digit is dialed instead of the configured no-key time out value configured. Also fixed deletion of an incorrect timer due to this action.

  • Fixed a string matching function which made our password checking case-insensitive
  • Fixed VLAN TCP issue
  • Added model number and firmware version (app) in every syslog message
  • Changed- In GUI Status menu item, "N/A" is displayed instead of "NO" for accounts configured to be not-in-use
  • Fixed the 3WC related problems by application restricting 1PCM+1LBR conference rule
  • Fixed under "Ring Volume" the volume is decreased by 1 each time you enter that menu item even if you did not make any change
  • Fixed we do not detect duplicate firmware (and fall into reboot loop) if the provisioning server responds first data frame < 512 bytes
  • Fixed we crash after receiving fragmented SIP messages introduced in 1.0.2.5

Bugs / Tweaks:

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!
  • MAJOR: (Mar31/06) Phone crashing The Phone crashes reproducable when more than eleven concurrent calls are sent to one account on the GXP2000 - Mitsch
  • MAJOR: (Apr18/06) Phone locked up The phone locked up when I was scanning through Missed Calls looking for a way to add the number to the phonebook - ADW
  • MAJOR: (Feb23/06) Rebooting. The phone seems to do a complete power cycle, including the ethernet switch, every 15-20 minutes. This has not happened in a call, but I don't see why it couldn't. - russlevy
    • NOTE: (Feb24/06) Check your ADVANCED SETTINGS -> Firmware Upgrade and Provisioning -> Automatic Upgrade settings. - bani
      • NOTE: (Feb25/06) I have already downgraded back to 1.0.2.8, since it was unusable in its other state. Automatic Upgrade was set to No, so that couldn't be the problem. - russlevy
        • NOTE: (Feb25/06) Do you have an old hardware revision? - mike240se
          • NOTE: (Feb26/06) No again :) - russlevy
    • NOTE: (Mar29/06) I have experienced reboots after 12-15 minutes while in calls every time, and it is very frustrating. It happened three times in the same conference call. I have never noticed the phone reboot while idle. This is major, since I'm trying to show a cheap imported phone will work as well as a Cisco 7960

  • MAJOR: (Feb24/06) Blanking Screen. Since installing 1.0.2.13, my phone's LCD becomes blank after about 10 seconds after a reboot, and the only way to fix it is to reboot. I've seen my screen display information for maybe 60 seconds total since I upgraded. - NateBell
    • NOTE: (Feb24/06) I can confirm screen blanking still occurs on phones with MAC addresses starting with 00.0B.82.03. The screens blank almost immediatly after reboot. Firmware seems ok on all other MAC addresses. I have this loaded on about 20 phones now testing. 5 of them are the older phones with MAC's that start with 00.0B.82.03. - phoneguy30
    • NOTE: (Feb25/06) I can also confirm its still affecting my old-MAC phone ... it goes blank after anything from 30 seconds to half an hour. once blank, it stays that way, it might still be corrupted , but hard to tell once its blank ;) . - robin_sz
    • NOTE: (Feb25/06) I'm not getting this on my old-MAC phone. The whole bisplay blank/scroll/corrupt has gone away on this beta, so far. However, it should be noted that I have the missed call list turned OFF. - jedi98
    • NOTE: (Feb27/06) Both my 00.0B.82.03.xx.xx test phones are still having the blanking problem, though it can be anyware from several seconds up to several hours before the screen goes blank... - SoloFlyer
    • NOTE: (Feb28/06) Screen blanking is not related tothe missed call list as i have this turned off on one of my test phones and it still has the problem - SoloFlyer
    • NOTE: (Mar11/06) Screen blanking has now cropped up again on my phone but was NOT triggered by an incoming call. The phone re-boots @ 6:30am (with a cron job), display was blank by 12:00 midday. This has happened 3 times so far, approx once every 3 days. - jedi98
    • NOTE: (Mar23/06) Time drags on .. still no update and my phone si still useless with a blank display ... any chance of grandstream offering swap-outs to those with the "problem" hardware for a newer phone? - robin_sz
    • NOTE: (Mar23/06) Keep in mind the entire 1.2 series is still technically a flash-at-your-own-risk beta. The first 1.2.x firmware didn't have the display bug as i recall, maybe use that for a while? I think GS is still crunching on the feedback they got from .13. Soon hopefully :) - Helix

  • MAJOR: (Feb24/06) Missed Call Lockups. The phone will lock up under some conditions while viewing and deleting items in the missed-call list. It appears that a power-cycle is required to get the phone back into a functional state. While I've observed this on multiple phones and multiple times on each, I have not found a guaranteed way of reproducing the bug. - thetatag

  • MAJOR: (Mar06/06) Random Lockups. Aside from those lockups described above under Missed Call Lockups, several GXP-2000's that I have tested will randomly lock up for no aparent reason, sometimes during a call, and sometimes while sitting idle. This is seperate from the rebooting / power-cycling issues discussed above, as the phone gives no indication that it has locked up, except that the clock stops advancing (nothing changes on the display) and sometimes the MWI light remains lit. In my test cases, this happens infrequently enough to not completely inhibit production usage, but often enough to be annoying — maybe every two to four days. - thetatag
    • NOTE: (Mar07/06) Are you able to ping the phone? Send a curl reboot? Are you using any special options? 1.0.2.13 i assume? - mike240se
    • NOTE: (Mar10/06) This may not be the same crash, but it sounds similar, though we get a crash maybe once or twice a month. We have a bunch of phones running 1.0.1.9 and every now and then one of the phones locks up. The phone can't be pinged, you can't access the phone's web interface. Hard crash. The MWI light turns on, and the display freezes. I noticed this tended to happen when a person was on the phone, and another person called the phone. When the caller got to the voicemail, the phone locked up. I turned off call waiting to see if that would help, and so far so good. Of course, this was with older firmware, so I don't know if the issue has been resolved in 1.0.2.13. -NateBell

  • MAJOR: (Feb06/06) qualify=yes in busy environments will sometimes cause 1-way audio when calls are resumed after being put on hold with MOH. (see asterisk-users post). a quick fix is to qualify=no and restart asterisk. - bani
    • NOTE: (Feb14/06) I have experienced similar condition with qualify = yes. When calls are resumed from MOH, caller is unable to hear our side. This only happens when there are many GXPs in use at the same time. I cannot reproduce condition unless our phones are active — cannot duplicate with just 1 SIP client/user. The only way I have been able to fix it is make qualify = no, and restart asterisk. - SamL
    • NOTE: (Mar23/06) I've just experienced the same problem on firmware 1.0.2.13 also It's only happening on 1 remote user going over multiple NAT's the other 10 with the same conditions are fine. - ChrisUK

  • MINOR: (Apr04/06) MWI doesn't work with STUN The Message Waiting Indicator doesn't seem to work with STUN - but does work with Outbound Proxy - crayfishuk

  • MINOR: (Mar28/06) Numbers off-screen. When entering long phone numbers (i.e. for international calls) the phone number goes off-screen as it advances, rather than moving up to the upper line as it used to. - Mike

  • MINOR: (Mar27/06) No Dialtone. Normally pressing a line button causes a speakerphone with dialtone. Today, pressing a line button gave no dialtone. In fact all dialtones had disappeared but the phone still worked. Rebooting fixed this & it has only happened once, hence it's a minor. - Jedi98

  • MINOR: (Mar21/06) Silent Ringing. When you call another SIP phone you don't get the "Ring-Ring" and is completly silent until the remote party picks up at which point it works as normal. Its completly random and happens often enough for me to be worried about putting it into a production environment. Any Ideas?. - ChrisUK
    • NOTE: (Mar23/06)If you're using Asterisk, add progressinband=yes in sip.conf, and you will allways hear the ringing. - mtryfoss

  • MINOR: (Mar13/06) Asterisk BLF. If the SIP account's username begins with "id", the BLF doesn't work: the phone subscribes, but BLF indicators don't light. It seems that it was not the case with the previous beta version. - snipefoo

  • MINOR: (Mar13/06) Voice Hic-ups. In the 1.0.2.13 firmware there is a 1 second hic-up / stutter at the start of the call when using full volume with PCMA and asterisk. This happened once with the first call after the firmware was first installed and has happened 2-3 times in a 4 day period and apears to be random. - ChrisUK

  • MINOR: (Feb27/06) Display Corruption - Artifacts. In the 00.0B.82.03.xx.xx phones there still appears to be a small amount of corruption left in the screen specifically in some of the decimal places , this was happeninging in the 1.0.2.8 firmware also but it is not evident in the photos that i had taken as the quality wasnt good enough.- SoloFlyer

  • MINOR: (Feb22/06) Display Issues. The 484 message when a time-out occurs partway through a number in early dial has been observed at least once as being displayed as "484ÿ". Grandstream has observed “503EE" on the display. This suggests that there may be a NULL termination issue or a pointer issue. Alternatively, there may be an issue with clearling the display buffer. However, reliably reproducing this minor bug has proven to be difficult. Please post an additional note if you discover a way to force this bug to manifest itself reliably.- thetatag
    • NOTE: (Feb22/06) easy to reproduce. exten => 5555,1,Congestion will do it. Apparently a response of SIP/2.0 503 Service Unavailable causes the gxp2000 to print 503EF - bani

  • MINOR: (Feb24/06) Early Dial / Transfer. When using the handset (not speakerphone) and early dial, after completing a blind transfer, the GXP-2000 goes back to a dial-tone and suggests that you can dial out again. However, the buffer that stores the early-dial number is not cleared, so the first digit you dial will be sent with the transfer number prepended to it. This is true even if you press another line button before dialing. (However, if you hang up the handset and pick it up again, the dialing begins fresh. Grandstream has acknowledged this as a bug a few versions ago, but has not yet corrected it. - thetatag

  • MINOR: (Feb24/06) Muted DTMF. You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag

  • MINOR: (Feb24/06) Paging Rings. Calling the phone with the SIP header "Call-Info: answer-after=0" will not cause the phone to answer immediately. The phone produces about half of a ring and then answers. At 0 there should be no ring, at 1 there should be one ring, etc. Grandstream has acknowledged this as a bug several versions ago, but has not yet corrected it. - thetatag
    • NOTE: (Feb27/06) answering after 0 rings would cause the recipent to have no indication that anything had happened, allowing for abuse... I suspect that this is not a bug but a deliberate feature ( or its a bug which they have chosen not to correct for the aforementioned reason ) a true answer after 0 rings should be added as a configuration option but it should not be enabled by default - SoloFlyer
      • NOTE: (Mar1/06) I agree that the half ring is very nice to have, if they do fix this so called "bug" i would also like to see the option to turn on the half ring. Without, it could be used to secretly monitor peoples conversations. - mike240se

  • MINOR: (Feb24/06) Asterisk BLF. If Account "SIP Registration" is "no" (ie. sip.conf uses host=xx.xx.xx.xx, not host=dynamic) BLF does not subscribe. - lonnie

  • MINOR: (Feb14/06) gxp2000 gsm codec is broken with asterisk music on hold. ulaw and g729 work fine, only gsm is broken. - bani
    • NOTE: (Feb17/06) same here - i think this has to do with silence suppression - when using GSM, silence suppression is obiously on (or it appears to be) even though the feature is turned off in the options - Adrian2k
    • NOTE: (Feb27/06) I had a call today using gsm and could not use the transfer button or the conference button. When trying to recreate it happened everytime. Anyone else haveing this issue? - hminc

  • MINOR: (Feb27/06) Ring volume configuration doesn't survive to a reboot. - snipefoo

  • TWEAK: (Feb22/06) Do Not Disturb. There is no reason for the Do Not Disturb menu to offer enabling and disabling DND mode. One of the options is already the current state. Only the other state and "Cancel" should be offered. Perhaps it should also tell you the current state (a dot next to the current mode, almost check-box style). There are also far too many button presses required to turn DND on and off since the GUI menu was introduced. - thetatag
    • NOTE: (Feb22/06) Probably a better way would be to just use MUTE/DEL as a DND toggle on the idle screen. - bani
    • NOTE: (Feb23/06) Now the mute/dell makes loads of sense because the DND is something you just want to press & go (out for lunch, coffee, snooze etc...) also it's intuative. - jedi98

  • TWEAK: (Feb24/06) Speakerphone volume setting should be independent of handset volume. I need to turn the speakerphone up to full volume to hear it, but this means the handset is too loud. - bani

  • TWEAK: (Feb24/06) Turn off speaker on remote disconnect should default to Yes - bani

  • TWEAK: (Feb25/06) Missed Calls. Its still painfully difficult to get rid of the missed calls notification. Hitting the up arrow should be enough to remove the notification. Or atleast make it a choice in the web gui. Having to either look at every call or scroll all the way to the bottom of the list to clear the list is too much work, especially if there are like 75 calls on the list. Yes you can turn off missed calls completely, but id rather have them. - mike240se

  • TWEAK: (Mar06/06) BLF / Speed Dial. When you press a non-BLF speed-dial button without being on a line, the speakerphone engages and the call progresses. This is appropriate behavior. However, if the speed-dial button is set to BLF, you can still call it, but you have to pick up a line first. It will not automatically engage the speakerphone. It seems to me that the BLF buttons should behave exactly the same as the speed-dial-only buttons, except with added BLF indication. (I have repeatedly asked Grandstream to change the drop-down selection of "Speed Dial" or "Asterisk BLF" to a checkbox to enable or disable BLF, so as not to suggest that it loses its speed-dial capabilities when using BLF.) - thetatag
    • NOTE: (Mar10/06) When the phone is IDLE and I press a speed-dial or a BLF button, the speakerphone engages (is it due to a particular setting ?). I think it would help some users if the speed-dial/BLF buttons were able to take a new line and place the active line 'On Hold', when you're already on line (to make easier the call transfer process) - snipefoo

  • TWEAK: (Mar18/06) Inverted colors Why the heck is the menu "inverted"... The whole office here agrees, that this is plain ugly... Blue on White with a more appealing slider would make it look far more professional... IMHO - KampfCaspar

  • TWEAK: (Mar18/06) BLF pickup in dialled numbers Picking up a BLF key effectively dials "**[extension number]". This process should NOT be listed in "dialled numbers" as no user interprets it as dialling anything. - KampfCaspar

(In addition to adding user feedback here, please move items from the 1.0.2.8 bug/tweak list to this list if you discover that they are still issues! Thank you! - thetatag)





Firmware Notes (Beta 1.0.2.8):


The firmware 1.0.2.8 is also attached to this page:
Image

The template for the GXP2000(w/ updated firmware) is here : http://aseclub.net/bigb/gxp2000/gxp2000_config_1 0 2 3.txt

Changelog
Build 1.0.2.8 2/6/2006

  • Fixed memory error syslogs not including right source

Build 1.0.2.7 2/3/2006

  • Fixed under "Ring Volume" the volume is decreased by 1 each time you enter that menu item even if you did not make any change
  • Fixed we do not detect duplicate firmware (and fall into reboot loop) if the provisioning server responds first data frame < 512 bytes
  • Fixed we crash after receiving fragmented SIP messages introduced in 1.0.2.5

Bugs / Tweaks:


  • MAJOR: Display Corruption Blanking, Wrapping, Upside down, and Artifacts... - SoloFlyer
    • NOTE: (Feb??/06) Display blanking problem from 1.0.2.3 still present where the display will clear completely until re-boot following an incoming (missed?) call. Display stays lit but blank. Also has been triggered by calling loopback call to the phone. This happens most but not every every call. There is a delay between the incoming call and the clear screen. - jedi98
    • NOTE: (Feb??/06) When using Speakerphone, display will often wrap so the top line (status icons/clock) are on the bottom. Due to some other oddness and that the divider below them is still there at the top, I'm assuming this is a bug. - Helix
    • NOTE: (Feb06/06) I can't reproduce this. Can you attach a screenshot of the bug? - bani
    • NOTE: (Feb07/06) Upon further research it's not a sp-phone bug, it just got triggered when I hit SPKR. This is the same bug discussed above that made beta pull this ver off his TFTP, my MAC addy is 00.0B.82.03.xx.xx so I am affected. See: here if you still want a screenshot. -Helix
    • NOTE: (Feb07/06) I can confirm this backward text bug still exists in version 1.0.2.8. I am using a phone with a MAC starting with 00.0B.82.03 as well. -Flu
    • NOTE: (Feb16/06) I can confirm, Screen Wrapping, Screen Artifiacts, Screen Blanking, and Screen upside down... Both the phones i am testing with have 00.0B.82.03.xx.xx MAC addresses.... Images: Upsidedown, Upsidedown and Corrupted Upsidedown and wraped, Wraped - SoloFlyer

  • MAJOR: (Feb06/06) qualify=yes in busy environments will sometimes cause 1-way audio when calls are resumed after being put on hold with MOH. (see asterisk-users post). a quick fix is to qualify=no and restart asterisk. - bani
    • NOTE: (Feb14/06) I have experienced similar condition with qualify = yes. When calls are resumed from MOH, caller is unable to hear our side. This only happens when there are many GXPs in use at the same time. I cannot reproduce condition unless our phones are active — cannot duplicate with just 1 SIP client/user. The only way I have been able to fix it is make qualify = no, and restart asterisk. - SamL
    • NOTE: (Mar23/06) I've just experienced the same problem on firmware 1.0.2.13 also It's only happening on 1 remote user going over multiple NAT's the other 10 with the same conditions are fine. - ChrisUK

  • MINOR: The ring volume setting doesn't survive to a reboot. -Adrian2k

  • MINOR: Loopback call (dial phone from itself, via asterisk) drops out on answer (expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up/down arrow adjust volume instead of missed call list. -Jedi98

  • MINOR: The ringtone selection (via display menu, but not via Web GUI) need a reboot to operate (that wasn't the case in the 1.0.1.x versions). - Snipefoo

  • MINOR: (Feb11/06) In Early Dial mode, the Delete button doesn't delete the last pressed digit from the screen. To reproduce, set phone to early dial mode, start dialing a number, then press delete. The deleted number(s) won't go away until a new number is pressed. - Ted

  • MINOR: You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag

  • MINOR: (Feb14/06) Phone displays 503EF when it should display Busy instead. - bani

  • MINOR: (Feb14/06) While phone is ringing, if you press UP, DOWN, or MENU key and then exit menu, phone rings with blank display. - bani
    • NOTE: (Feb20/06) Its not actually blank it still shows the bar at the top, but that is all it shows it goes back to normal after the phone stops ringing though... -SoloFlyer

  • MINOR: (Feb14/06) gxp2000 gsm codec is broken with asterisk music on hold. ulaw and g729 work fine, only gsm is broken. - bani
    • NOTE: (Feb17/06) same here - i think this has to do with silence suppression - when using GSM, silence suppression is obiously on (or it appears to be) even though the feature is turned off in the options - Adrian2k

  • MINOR: (Feb17/06) BLF lights stay on after the phone looses the connection to the Asterisk box - Adrian2k

  • MINOR: (Feb19/06) I have 2 registered accounts on Line 1 and Line 2. Line 3 has wrong settings, so it can not register and shows "Not Registered" when the phone is lifted up. When I press "LINE 2" and back to "LINE 3" the message "Not Registered" disappeared. In reality, it's still not registered. - PaulK

  • MINOR: (Feb21/06) When the phone has no ip assigned (network unplugged) you can't access the phones menu system, hitting the round key does nothing. For example, i needed to change the registration config so it wouldnt conflict with a phone already on the network, but couldnt. - Mike240se

  • MINOR: (Feb22/06) I phoned someone over the phonebook and didn't lift up the handset. When he hang up the phone my GXP2000 began beeping the busy-noise for a whole minute (or more?). It would be better to stop the busy-noise after 3 seconds. - PaulK
    • NOTE: (Feb24/06) I don't think this is a bug. Try Account 1-4 -> Turn off speaker on remote disconnect -> Yes. If this fixes the problem for you, you should remove this from the buglist. - bani
    • NOTE: (Mar05/06) If that is the case, then yes should be checked by default (i just noticed, you stated it already above). Why would you want to hear the busy-tone all the time? Anyways, I gonna check if that fixes it. - PaulK

  • TWEAK: (Feb19/06) (issue from 1.0.2.3) There's still a spelling error somewhere: "not alloweded"- PaulK

  • TWEAK: (Feb19/06) (issue from 1.0.2.3) It should be enough to see into the "Missed Calls" list to let the "1 Missed Calls Press /\ key to ..." frontal remindermessage disappear.- PaulK
    • NOTE:(Feb21/06) I agree this is extremely annoying, there should atleast be a setting in the web interface where you can decide how you want this handled. - mike240se

  • TWEAK: (Feb13/06) On Missed Calls list, RIGHT button should jump to Clear All, and UP button at the top of the list should jump to the bottom of the list.- bani
    • NOTE: (Feb16/06) Fundamentally, it seems, the main problem with missed call is there's no "new missed calls" concept. How about del key clears the screen message so we can see the real screen & turn the light off? Config option to NOT show screen message because it's a pain in ringing groups of phones. - Jedi98

  • TWEAK: Speakerphone volume setting should be independent of handset volume. I need to turn the speakerphone up to full volume to hear it, but this means the handset is way too loud. - bani

  • TWEAK: (Feb15/06) Config->Factory Reset is too cumbersome to use. 2 or 3 levels of "are you sure?" would be sufficient, but requiring the entire MAC address is way over the top — that is like 12 levels of "are you sure?". This makes it incredibly cumbersome when you need to factory reset dozens of phones. - bani
    • NOTE (Feb21/06) There should be a way to factory reset when plugging in the phone, like plug in the phone while holding 2 keys down at the same time. - mike240se





Firmware Notes (GXP-2000 firmware v1.0.2.6)


Changelog
Build 1.0.2.6 2/2/2006

  • Fixed GUI sometimes send additional key events (cause of the SIP configuration problem). Now the focus is default at the "CANCEL" button in each SIP edit dialogs. This also resolves the Upgrade menu inaccessible problem
  • Fixed GXP displays "TFTP Provisioning" briefly before correcting it to "HTTP Provisioning" if HTTP is in use
  • Fixed ring tone file problems
  • AGC change
  • Changed- FUNC keys would act as speed dials even when BLF status is BUSY
  • Changed- All keys are blocked when provisioning is in progress (including LINE keys and SPEAKER key)
  • Changed- Factory reset will now clear phone book as well as custom ring tones
  • Changed- GUI->Config->Upgrade changed to allow edit firmware and config server (the original interface was implemented before the application TFTP), Note that the old 12 digit IP address format you use for direct IP calling is no longer valid here. You will need to type in the dots (* key) to separate the octets.
  • Fixed we fall into reboot loop when there exists a ring3.bin in the server and ring2.bin spans over 64k
  • Fixed we do not reboot immediately upon receiving a cfg file (that caused a change)
  • Changed- we no longer reboot if only ringx.bin are downloaded
  • Fixed we display line status as MUTE if the previous call is in MUTE and disconnected by remote party
  • Fixed when account name length > 16 characters, GXP-2000 freezes after a call (Bugzilla #135).
  • Added allow user to use the Speed Dial keys to do blind transfer
  • Fixed we do not process the IP packets if the first fragment did not arrive first
  • Fixed we still display AM/PM even when 24 hour display mode is selected
  • Fixed our LCD backlight does not light up immediately when a call comes in
  • Added- LCD backlight stays light up when there are unviewed missed calls to alert user, the LCD backlight also stays up whenever the MENU operation is in progress
  • Fixed GUI SIP configurations were not accessible
  • Changed- GUI menu item sequence rearranged for user-accessibility, Status page items rearranged for a quick glance at registration status
  • Changed- when display mode is set to DDMMYYYY and reverse date/time is set to YES use period as separator instead of hyphen (I have seen many Europeans request for this and this does make sense so that when you see a date 03.04.2006 you can tell if 03 is the month or 04)
  • Changed- when display mode is set to DDMMYYYY and reverse date/time is set to NO, the long date string is displayed as "dddd, dd MMMM" where "dddd" is day-of-week in English, "dd" is day-of-month in number, and "MMMM" is month in English, example: "Friday, 27 January" (standard mode is "dddd, MMMM dd")
  • Added allow disable miss-call features as per-account setting, changing this setting takes immediate effect without reboot. Incoming call is still logged, only missed calls are not. It can be provisioned using P182/442/542/642, valid values 0 and 1, default is 0 (No) which WILL log all missed calls.
  • Added disallow MENU actions when provisioning is in progress
  • Changed-when you use the UP arrow key to view missed calls, you will return to main idle screen directly if you either use the LEFT key or delete all missed calls
  • Changes to phonebook: on the main Phone Book menu, the "Add" is renamed to "New Entry", in the New Entry page the "Add" is renamed to "Confirm Add", "Back" is renamed to "Cancel & Return"
  • Fixed phonebook entries remains in flash (and reappear after reboot) after delete

Bugs / Concerns:


  • MAJOR: Endless http upgrade reboot bug is still there. Seems the GXP2000 is confused by the http responses returned by Apache 2.0.53/54 on Fedora Core 4. Anyone can reproduce the bug by pointing your phone's Firmware Server Path to mihoshi.anime.net/gsexperiment at your own risk! do not do this unless you are able to recover from an endless upgrade reboot cycle on your own. i will not be around to help you recover your phone if you cannot. - bani
BUG FIXED IN 1.0.2.7 ALPHA. NEXT BETA RELEASE SHOULD NOT HAVE THIS BUG. - thetatag

  • MINOR: Everytime you go into the ring volume setup, the volume decrease of one unit. - Snipefoo
BUG FIXED IN 1.0.2.7 ALPHA. NEXT BETA RELEASE SHOULD NOT HAVE THIS BUG. - thetatag




Firmware Notes (GXP-2000 firmware v1.0.2.3)


As of 1/26/2006, Grandstream has officially released for Beta testing GXP-2000 firmware version 1.0.2.3. Following the very successful format of detailing the last firmware beta, the following lists features implements, bugs, etc. For complete release details, see Release Notes. - thetatag

GXP-2000 firmware is larger than other Grandstream product firmware and downloading time is normally 15 minutes.
Also, there are major changes in firmware 1.0.2.3 which requires firmware to be downloaded twice and power cycle as well
- DigiGuru

WARNING:
(Feb01/06) Grandstream has stated that there is currently no way to downgrade from the 1.0.2.x to the 1.0.1.x series. If you upgrade to the 1.0.2.x firmware, you are stuck with it. Grandstream has indicated that a transition file may be available at some point which would allow such a downgrade of firmware, but it is not available now. If you choose to participate in the beta, you have been warned! (Fortunately, this version seems pretty good for the most part, so I doubt too many people would regret moving in this direction, especially if you have a non-production environment in which to test it.) - thetatag

Changelog
Build 1.0.2.3 1/24/2006

  • Added GUI Interface
  • Added Address Book
  • Added Call Log
  • Shortened the wait-time between downloading files from 1 second to 100ms
  • Added configurable T2 interval--this is a per-account setting. Possible values are 2/4/8 sec. Provision parameter P250/P441/541/641, valid values 200/400/800 (in 10ms unit). Invalid values ignored, default value 4 second.
  • Fixed display fonts not aligned correctly during call-time (the call-timer ticker should align to the right of screen)
  • Fixed miss call count does not display the 0 at the end
  • Changed max call log to 50 entries
  • Added configurable T1 timeout interval--this is a per-account setting. Possible values are 0.5sec/1sec/2sec. Provision parameter P209 (P440/540/640 for accounts 2-4 on GXP-2000), valid values 50/100/200 (in 10ms unit). Invalid values ignored, default value 1 second.
  • Added under Broadsoft mode, DNS SRV fail-over happens after 3 retries (so if you set T1=0.5 sec, it takes 7.5 seconds to fail-over to second server).
  • Fixed session-timer does not work properly before session establishes (UPDATE/481 issue reported)
  • Fixed we do use increment CSeq in the INVITE to unhold--this is a general 100rel bug what will occur whenever 100rel (PRACK) is enabled and impacts GXP-2000 as well.
  • Added LCD displays provisioning status and warning message when flash writing/erasing is in progress
  • Added provisioning protection- During provisioning all incoming SIP packets will be dropped without processing
  • Added implementation of T2 timer (see RFC3261) and send BYE when 200 OK time-out
  • Added we will do TFTP/HTTP provision upon DHCP/PPPoE completion if no IP address was available initially
  • Fixed VLAN bug
  • Added TCP/HTTP fix
  • Fixed TFTP retry issue
  • Fixed we respond 200 OK with "event: presencenoevent" for the SUBSCRIBE (event: presence) we receive
  • Fixed when reversed-date/time selected and DD-MM-YYYY is selected, the date display is not displayed properly

Features / Fixes

  • FEATURE: Menu. When the phone is on-hook, the round button brings up a simple text menu (complete with scrollbars) giving access to various configuration options. It is much more useful than the old-style configuration and gives that much more credence to the GXP-2000 as an enterprise-grade phone. - thetatag For unknown reasons, the menu system uses inverted colors which looks strange. Most functions now requires more key presses to access. No menu selection using the keypad. Uses OK and Cancel buttons for actions which has no corresponding keys, you have to access them by combinations of round key and arrow keys (depending on where cursor is positioned, which isn't obvious), making it hard to navigate. - job
  • FEATURE: Phone Book. The phone book is only partially useful at this point. You can manually add entries to it and use it to dial those entries, but there is currently not support for grabbing a number from the call logs (either incoming or outgoing) and placing them into the phone book. So while it might be slightly useful, be aware that you'll have to enter everything manually, and it doesn't appear that you can use configuration files to include complete phone books in your provisioning. Also note that you cannot currently assign phone book entries to any sort of speed dialing. They stand alone. You go into the phone book if you want to use it. Also, it does not use the name associated with a number in your phone book for CallerID. - thetatag
  • FEATURE: Missed Call List. When the phone rings and you don't answer it, the screen displays, for example, "1 New Missed Call" "Press ^ Key to view". Pressing the up arrow brings up a list of missed calls, including time/date stamping, account number, and callerID information. It also gives you the opportunity to redial one of these entries. Currently there is no way to enable or disable this feature on an account-by-account basis, so if you use a seperate account for all of your internal calls (for distinctive ringing or whatnot), you cannot keep these numbers from showing up in the list even if you don't care to track missed internal calls. - thetatag
  • FEATURE: Do Not Disturb Icon. The "Do Not Disturb" mode now has a nice flashing icon at the top of the display, rather than announcing itself in place of your time or date. Regretably, there is still no account-by-account DND mode. In addition, the DND option is now under "Preferences" in the menu, requiring several more button presses to turn it on and off. - thetatag
  • FEATURE: Paging Support. If you pick up a line (so the phone says "DIAL USING") and press the round button before dialing, the prompt changes to "PAGE USING". This simply inserts the SIP header for auto-answer support. If you're doing anything other than peer-to-peer (which doesn't support the page option, to my knowledge), your PBX will need to look for that SIP header to do anything useful with it. - thetatag
  • FEATURE: NAT/Router Support. This firmware now allows the GXP-2000 to act as a Network Address Translation router, with the LAN port being the "outside" and the PC port connecting to the internal network. Through the web interface the GXP-2000 can be configured for NAT or traditional bridging (the behavior of prior versions). There is currently no support for non-NAT routing. - thetatag
  • MINOR BUG FIX: The problem in 1.0.1.13 in which all codecs did not appear for every account has been corrected. There are currently 5 supported codecs (a disappointment for those of us hoping to see SPEEX): GSM, G.723.1, G.729A/B, PCMU, and PCMA. Why there are eight options in the preference list is still a mystery. - thetatag
  • MINOR BUG FIX: With the new menu system, you no longer have to pick up a line to use the recent callers and recently called lists. In fact, just the opposite is true. There is no longer any way to access these lists while off-hook. - thetatag

Bugs / Concerns

  • MAJOR: No longer possible to do conference calls with latest beta firmware. - Michael
    • NOTE: (Feb01/06) Michael: I have tested this on my GXP-2000s and have not had any loss of functionality. (I'm not willing to commit to it working perfectly even on mine!) Could you explain in a little more detail exactly what the phone does or doesn't do when you try to use the conference function? - thetatag
    • NOTE:(Feb3/06) I have just set up my third GXP-2000.with the latest beta firmware. Conferencing works perfectly on all three phones. - emckinnon
  • MAJOR: Both tested phones died (bridged network). Soft reset doesnt bring network back, only hard restart solved this issue (no dhcp, static IP, network LEDs do nothing). - Festr
  • MAJOR: Lines can't be called when BLF status is busy - most of my extensions are capable of muliple simultanious connectons. - Adrian2k
  • MAJOR: The phone goes into an endless loop with firmware upgrades on this version. After the phone reboots for the second time (after fetching boot55a.bin and gxp2000a.bin) you need to rename the files so the gxp can't grab them again. Otherwise it will just keep grabbing the same firmware over and over again. - bani
    • NOTE: (Jan27/06) Bani: I have been in touch with Grandstream concerning this bug report. Please provide some additional information. What firmware version were you upgrading from? Which upgrade method did you use? Did you completely replace all of the files for the upgrade (including boot55.bin)? Are you using any custom ringtones? If so, how many and what are their sizes? Please verify that none are larger than 64K as well and that none are the old BT100 ring tones. I have upgraded several phones without being able to recreate this problem, so any additional information would help diagnose and correct it. You can contact me at admin {at} composite-tech {dot} com. Thank you. - thetatag
      • NOTE: (Jan30/06) thetatag: happens via http updates. no custom ringtones at all. i am 100% certain the bin files are correct. other users on #asterisk have been able to reproduce the problem. - bani
    • NOTE:(Feb3/06) The first two GXP-2000 phones that I upgraded via tftp had this problem. I believe it relates to BT102 stock ringtones in the tftp directory. I renamed the BT102 ringtone files and added three custom ringtone files created with the tools available from the Grandstream web site. The third upgrade via tftp succeeded the first time. The file, ring3.bin does not load. Files ring1.bin and ring2.bin do. Its not the ring3.bin file as swapping 2 and 3 allows the former ring3.bin to load. - emckinnon
  • MAJOR: The display went blank completely after a period of about one day and would not show anything until a re-boot. After another day running (idle) the display blanked again. Phone previously had v1.0.1.13 firmware. - jedi98
    • NOTE: (Jan27/06) Since the beta has only been out for a day and a half, obviously it's too soon to tell if this is a freak incident (static electricity, swamp gas, sun-spots, etc.) or a major firmware issue. But if it is a firmware problem, it's definately worth making Grandstream aware of. Were you using the phone when this happened? Has it happened to multiple phones? Also, if anyone else reading this has had this problem, please indicate such and provide some basic information on any phone-related events leading up to the crash. I will present any information this community can put together to my technical contact at Grandstream. Thank you. - thetatag
    • NOTE:(Jan27/06) I have encountered this bug as well. I am testing the new firmware on 2 phones right now, and this morning when I got to work, one had a blank screen. It still seemed to be functional, but I could only restore the screen by reseting the phone. The other phone is doing fine. I don't know what the conditions were when the screen blanked. All I know is it happened over night, so I doubt anything out of the ordinary occured. I'll update if it happens again over the weekend. NateBell
    • NOTE:(Jan30/06) This bug same for me, tested two phones . Festr
    • NOTE:(Jan30/06) Over the weekend the screen on the same phone as before turned off. Curiously, the other phone I'm testing with does not have this bug. The only difference between these phones (besides differing SIP numbers) is the format for the date/time. The phone with the blank screen had the time displayed in the center while the other has the date in the center. Both phones still seem to operate fine. NateBell
    • NOTE:(Feb3/06) Two of my GXP-2000 have been running six days without this problem. - emckinnon
  • MINOR: It seems the phone no longer lights up when receiving a call. Seems a little strange, and this IS a beta firmware. If this is going to affect you, do not upgrade. I'm looking for an option to turn this back on - DigiGuru
    • NOTE: (Jan27/06) Grandstream has acknowledged that the fact that the phone waits until the second or third ring to light up is a timing bug and has assured me that this will be corrected prior to the production release. - thetatag
  • MINOR: The display shows AM/PM-indicator even when clock display is set to 24h-style.
  • MINOR: When adding a phonebook entry, the backlight sometimes switches off and gets switched on with the next typing of a letter, timeout-counter doesn't get a refresh by typing? - PaulK
  • MINOR: There's still a spelling error somewhere: "not alloweded" - PaulK
  • MINOR: If you hang up on a call while it is muted, the next call (either incoming or outdoing) on that line will also be muted. The flag to maintain the mute status for each line needs to reset when the call on that line terminates. - thetatag
  • MINOR: When a dial plan prefix is configured for an account it is not used when dialing from the call history directory. - Nezer
  • TWEAK: It should be enough to see into the "Missed Calls" list to let the "1 Missed Calls Press /\ to ..." frontal remindermessage disappear. - PaulK
    • NOTE:(Jan30/06) Press the center button one more time when drilling down. Although, I agree that bringing up the list should clear the message. Nezer
  • TWEAK: The "Missed Calls" list should try to resolve the numbers to names.- PaulK
    • NOTE:(Jan30/06) Press the center button and the CLID information will be in that screen. Nezer
    • NOTE:(Jan31/06) Imho, that's too late: The list should already resolve it. - PaulK
  • TWEAK: edit number allows for letters too, so if you have a number with two "1" in a row it will print a space instead of "11" .. it should allow numbers only. - PaulK
    • NOTE:(Jan30/06) What about a number that's a direct dial IP call (some_sip_address@voipprovider.net)? Nezer
    • NOTE:(Jan31/06) Hm, a way to distinguish between them would be nice. - PaulK
  • TWEAK: it's not possible to delete letters and numbers from phonebook entries, number, name etc... a delete key is missing! - PaulK
    • NOTE:(Jan30/06) Yes it is although it's not intuituve. Use the right/left cursor keys and go to the character immediatly before the one you want to delete and then press the 'mute/del' key. Nezer
  • TWEAK: an "Exit" menupoint is missing in the "Missed Calls" details of a number to get out of the GUI. Instead you have to lift the phone and down to leave the menu quickly. - PaulK
    • NOTE:(Jan30/06) Again, not really intuitive but keep pressing the left arrow key and you will return to the main display. Nezer



Firmware Notes (GXP-2000 firmware v1.0.1.13)


  • Handset volume issue - FIXED (Thankyou GS!)
  • Real Intercom, yeah baby no more workaround :) Use SIPAddHeader(Call-Info: answer-after=0)
  • Ring volume adjustment issue (adjusting ring tone using up/down doesnt appear to work (it still works if you go to the menu item it just doesnt work on the non-menu screen)).
  • New Codecs
  • some things (speed dial etc) rearranged on web interface
  • Asterisk BLF support for speed-dial buttons - oh yeah!
  • Option to hang up on disconnect (for those who were annoyed with the busy signal previously generated)
  • No more raspy DTMF when dialing
  • Now you can dial rapidly without the phone missing digits
  • Less annoying call waiting beeps

And now for the flip-side of things:

  • MAJOR: Heavy, loud, ugly echo on handset - intermittant
    • NOTE: (Nov23/05) Additional research and testing suggests that back in 1.0.1.9 and possibly 1.0.1.12 this problem was directly related to the handset volume, and higher volumes created the dreadful echo. This may be the case in 1.0.1.13 as well, although users have reported that 1.0.1.13 provided worse handset results than earlier versions. Please fix this, Grandstream! - thetatag

    • NOTE: (Nov25/05) I can confirm that this is a problem, it appears to be caused by the the volume being increased to the point where the handset mic receives what is being transmitted from the handset speaker , worst case scenario happens when two people with volume at maximum talk to each other... within a few seconds both phones will be screaming with feedback it has been a problem in every version of firmware grandstream have produced but due to a volume bug in 1.0.1.12 where maximum volume was equivilent to a setting of about 50% (unless you followed the workaround) it was more difficult to experience problems with feedback... im not sure about the best solution to this problem but it may be possible to use the "Adaptive Digital" echo canceller to solve it (without reducing the maximum receive volume) the other optioin may be to keep the microphone sensitivity setting slightly lower when on maximum volume... - SoloFlyer

    • NOTE: (Nov29/05) Further extensive testing shows cases where this problems is much worse on 1.0.1.13 than 1.0.1.9. (1.0.1.12 is just too buggy to conduct live testing.) In some cases, even decreasing the handset volume all the way did not remove or even quiet the echo. - thetatag

    • NOTE: (Dec16/05) Additional testing has suggested a work-around that is working well on one particular system and may work well for others as well while Grandstream corrects the problem in their phones. See Grandstream GXP-2000 - Solving Echo Problems for details. - thetatag

  • MAJOR: Layer 3 QoS in the "ADVANCED SETTINGS" menu has no effect. Confirmed with tcpdump. tos is always 0x48 regardless of the setting in the menu. - bani

  • MINOR: BLF remains on if a call originated from an extension where a distinctive ringtone is set.
  • MINOR: Speed dial buttons in BLF mode are disabled when the extension is in use or the BLF subscription failed! (Why?!?)
  • MINOR: After pressing TRNF, the audio cuts out and you get a nice dialtone. But after you dial the first digit, the audio stream still kicks back in until you complete the transfer.
  • MINOR: Speed dial buttons still do not work for blind transfer or 3-way conference. Properly implemented, they should do just that: speed dial — as in, behave as though you quickly dialed those digits - even if it isn't a complete number. Come on, Grandstream. Really.
  • MINOR: The HTTP configuration lets you set the No Key Entry Timeout. However, when using Early Dial, this timer is still not used. Instead an approx. 3-second timer is used and cannot be configured.
  • MINOR: Same as the last bug report, regarding the No Key Entery Timeout. This also affects transfers with the TRNF button. And because of the bug about audio reconnecting after the first digit, there is no way to know when you're still transferring the call and when it has timed out and reconnected you to the other party who is now getting all of your DTMF in their ear. Some prompt regarding transfers should remain on the screen.
  • MINOR: Phone is still prone to lockup when rebooting after firmware upgrade.
  • MINOR: Ring Volume adjustment can still only be changed though menu (it used to be the default if you just pressed up and down... please bring it back)
  • MINOR: BLF shows extension in use when the state is Unavailable.
  • MINOR: While BLF is mostly functional, it is listed as Asterisk BLF, so one would think that it should not generate warnings on the Asterisk system. Under some valid conditions and proper configuration, Asterisk reports Incoming call: Got SIP response 415 "Unacceptable Content-Type" back from xxx.xxx.xxx.xxx when BLF-related events occur.
    • NOTE: (Nov29/05) When receiving the "''Incoming call: Got SIP response 415 "Unacceptable Content-Type" error, the BLF doesn't seem to work - looks like the phones are refusing the BLF messages all together. - conexim

  • MINOR: Under certain rare circumstances, BLF indications will be left "stuck" in an incorrect state. To counter this, the GXP-2000 may need to periodically drop its BLF subscriptions and renew them. Kind of a hack for what may be an Asterisk bug, but right now the only solution is to reset the phones.
  • MINOR: Rare, but spontaneous lockups when phone is idle. (Not PoE related.) Maybe Grandstream should consider a watchdog timer that can restart modules without a full reset if something fails or hangs.
    • NOTE: (Nov29/05) Calls intermittently freeze (audio completely muted), line light stays on (blocked), and cannot be stopped by pressing any button - the call eventually times out in asterisk. - conexim
  • MINOR: (Dec12/05) When VLAN tagging is enabled everything TCP related is now broken. You must do a factory reset to get access to the web interface or get the phone to accept anymore updates. (I dont have one of these phones but i've seen this happen on lots of other hardware - can you reduce the MTU size on the phone, if so reduce it a little and see if that helps, the extra bytes from the vlan tag have a nasty habit of messing some stuff up)
  • MINOR: Placing the phone on MUTE while on a conference call not only prevents the other parties from hearing you, but also from hearing each other. You can still hear everything, including their confusion. (This may only mute one of the remote parties in addition to yourself. Additional testing may be required.)
  • TWEAK: You still cannot blind transfer a held call. You either have to pick up the line first, or do an attended transfer. (Easy to work around, but blind transferring a held call would be really useful.)
  • TWEAK: A common question in an office deployment of GXP-2000's is "What does xxx.xxx.xxx.xxx mean?" Having the IP address always displayed is nice in some environments, but it should be optional.
  • TWEAK: A configuration option should exist to turn sidetone support on or off, and possibly to adjust its relative volume.
  • TWEAK: Pressing Send to complete transferring a call should be optional, and coupled with the Early Dial configuration option. If the technology does not allow this, at least the timeout should be used to determine when you are done dialing.
  • TWEAK: There is currently no way to see if the phone is forwarded when call forwarding is enabled locally.
  • TWEAK: When in a conference call, pressing one of the conferenced lines should immediately put the other on hold and connect just the pressed line. You should not have to quickly press hold and then the line you want to talk to.
  • TWEAK: When the phone is in Auto Answer or Auto Answer via Call Info with rings set to 0, the phone still produces a half-ring before picking up. Notification of the incoming page should be configurable or non-existant (leaving it up to a PBX). Configurable would be a better option as a PBX will not always be in the middle of some connections.



Firmware Notes (GXP-2000 firmware v1.0.1.12)


  • The automatic reboot after firmware upgrade appears to cause the phone to lockup... all lights inc backlight remain on and nothing appears on screen... hard restart ( disconnect reconnect power ) solves the problem and everything works perfect from then on - very annoying if upgrading multiple phones... Upgrading via HTTP didn't seem to cause this issue for at least one user.
  • Ring volume adjustment issue (adjusting ring tone using up/down doesnt appear to work (it still works if you go to the menu item it just doesnt work on the non-menu screen)).
  • Handset volume issue ( Sometimes the volume setting at full becomes equal to that at 3 or 4, making the handset very quiet. Doing an Audio LB test or switching to and from speakerphone seems to fix this problem temporarily)
  • Speakerphone is now usable with AEC. WOOHOO! :)
  • Still no intercom but the workaround works fine so who cares...



Firmware Notes (GXP-2000 firmware v1.0.1.9)


  • Speakerphone Suffers from signifigant Acoustic Echo
  • Still no real intercom system... I have been able to work around this by setting up a seperate user account on line4 which has auto answer enabled :)





Image



Due to the main GXP-2000 page becoming overcrowded with notes about old beta and other legacy firmware versions, that information has been moved to this page. - thetatag








Firmware Notes 1.1.5.15:

Release Notes:

Build 1.1.5.15 (11/19/2007)
• Fixed GXP display "Not Registered" for accounts that does not require registration
• Fixed GXP1200 LCD display corrupted
• Fixed AEC divergence during call waiting tone
• Fixed default router gets changed under static IP mode incorrectly
• Fixed GXP1200/GXP2010 out-of-memory problem/idle screen crash. Note: A hard power-cycle is REQUIRED after the new firmware is loaded.
• Fixed softkey problem with onhook dial
• Fixed softkey problem with direct IP call
• Fixed GXP2010 color LED blinking issue
• Fixed GXP2000 EXT issue
• Fixed BT200 redial problem after deleting called log. Fixed GXP missed call cannot be removed by LEFT key completely
• Added support for padding bit in RTP
• Fixed GXP2010/GXP2020 LCD backlight does not function correctly
• Fixed BT200 change codec from UI MENU will also change Voice Frames Per TX incorrectly


Post your experiences here:
  • Major : Phone crashes whilst on mains power, leading to situation where phone will only boot without a patch cable plugged in, has happened to 30% of our handsets within 18 months. Changing power and/or network cables does not fix the issue.- SummerIT
  • Major : Here's a new one for you to puzzle over... the phone is registered OK, shows the IP address, clock is correct, clock cursors flash, but when picking up handset, or pressing "speaker" there is no dial tone - but the speaker symbol appears at the top of the screen, I can enter numbers, they appear in the in the display, I can hear tones when pressing numbers, but pressing "send" or waiting for it to time out has no effect - the speaker symbol shows, but it doesn't dial. HV1.1A, FW1.1.5.15. PITA2.0 - olly
    • Confirmed. I'm also having this issue... EXTREMELY annoying, because people can't reach me, although for me everything seems all right. - mirak
    • Note : I have had this same issue with a GXP2010 running FW1.1.5.15. The phone still shows as registered to the PBX and receives MWI updates. It will not receive or make calls. When trying to place a call, the LCD screen shows some incorrect formatting, such as words written on top of one another. Seems to happen once the phone has been up for about three to four days. - rconner1890
  • Major : Why oh why do I try? Busy tone at exactly 2 minutes into the conversation, then disconnects at exactly 2.5 minutes. Nothing seems to make this piece o crap happy! :( - indesignfirm
  • Major : My phone (GXP2000) just rebooted in the middle of a conversation. - doesn't seem to be deterministic, but it's quite annoying, as my phone's in production use - and not test use :( - is the previous 'stable' version still available somewhere, so I may try and downgrade? Random reboots of the telephone are just unacceptable... :( - kodomo
    • Note : Me too! two out of five GXP2000 with this software restart at random, approx every 24 hours - olly
    • Note : Kodomo and Olly, please state the Hardware Version of your phones. I have seen that older HW versions seem to have a lot of issues with modern firmwares. Check hardware verison by entering phone menu, selecting "status", then scrolling all the way to the end and looking for a string that starts with "HV:" - Kurgan
      • Note : GXP 2000 HV1.1A - olly
      • Note : To me, it seems completely random (not even the 24h) I haven't been able to reproduce it yet - HV:v0.4 - kodomo
        • Note: Kodomo/Olly, please contact Grandstream for this, this issue should be addressed. Before that, you can try "gray"s magic, disable watchdog and see if it works
          • Question: How (may) do I disable watchdog permanently? - GMaoret
          • Note: After their spontaneous rebooting in days 1,2 and 3, my phones have not rebooted now for between 6 and 10 days - could it be that they download extra bits of firmware in the first few days, which requires a reboot? - olly
          • Note: For me it's the same, although I don't believe it's a download, since I disabled downloads again immediately after my update. I'm quite puzzled by this, but I've been unable to reproduce the reboot issue (even trying 2-3day long calls - incoming and outgoing)... I guess it's a rare race condition in my case(?) - kodomo
      • Note : HV:1.1A No problems with the call disconnection - nonsequitir
    • Note : Me too, reboots spontaneously on or off call ~once per day. disabling watchdog under factory settings seems to have an effect. HV:1.1A -p
  • Major : Two identical GXP-2000 + EXT stopped showing extension status on the EXT module. Transfers originating from the EXT module buttons where also triggering another extension to ring in addition to the desired extension. Don't know what triggered that condition but a reboot of the phones settled the problem. Both phones were up for 8+ days (since install) before this happened. Will wait another 8 days to see if this returns. HV is 1.2b - jflaplante
  • Major : GXP2000 (Hardware 1.1A) Call self-terminates after about 30 seconds when on HANDS FREE with MUTE enabled (Ticket#: 2007121199000214) - nonsequitir
    • Note: Grandstream support have recommended that I set "NAT Traversal (STUN)" as "No, but send keep-alive" - nonsequitir
    • Note: gray, you may want to check if the call gets established, from console, if still not working, check with Grandstream Support.
    • Note: Me too! This has been driving me nuts for as long as I've had this phone. I do a lot of conference calls, and inability to mute on speakerphone sucks. HV:0.4 - mjmac
  • Minor While this is not a new issue, it should be addressed. When logging into the web interface with the end user account, the Ext modules cannot be accessed. I believe setting button number options is an end user feature, not one for an admin. - bmullins
  • Feature Request : Add the ability to adjust microphone volume level. I thought this use to be a feature, maybe i got it confused with my snom 360 which allows this basic feature. Some gxp-2000 are way to loud to the other caller. - mike240se
  • Major : Language Pack 1.3 is unusable. Language file is named gxp2000.lpf instead, and if renamed to gxp.lpf the phone cannot write it. Phone's syslog shows: "ERROR write gxp.lpf" - Alex
  • Major : I have two GXP200s which have been rebooting quite often since I installed 1.1.5.15, mostly seemingly randomly during calls, but recently one started rebooting every time I woke my adjacent MacBook Pro from sleep, which I think I narrowed down to happening when the Airport wifi adapter was enabled and not otherwise. I've just disabled WDT and if that doesn't fix the rebooting I'm getting ready to throw the phones out. Any recommended replacements? adw 2008-02-03 Disabling WDT may have been a red herring. The ethernet out cable for connecting PCs seems to be a major contributor to the random reboots. Without that cable, the phones are less susceptible to random reboots, but still not completely reliable. And it's a pain not to be able to use that port, so still looking for a replacement adw added 2008-02-21
  • Major: After upgrading to 1.1.5.15 I have a few different issues. We are experiencing call dropouts after a couple of minutes (crackle/screech then dropout) MinorOn the interface, I cannot change optional rule in ntp, it always comes back to the default - annoying as it means I cannot make it display the correct time for South Australia - even though it is configured by both DHCP and NTP. Also I cannot get it to always use the lcd backlight. again, it just wont change the setting via the interface tmcruisin 04-03-2008






Firmware Notes 1.1.5.10 (Beta):

Release Notes:

Build 1.1.5.10 (11/01/2007)
• Fixed audio stack overflow crash (A1000 crash problem found on GXP1200)
• Fixed GXP2010/2020 LCD contrast MENU OK/CANCEL soft key
• Fixed onhook-dialing account selection bug
• Fixed GXP1200/2010 TFTP display incorrect model name and remove hyphen in all model names
• Adjusted MWI flash interval to 1 second between on/off
• Removed eventlist URI settings from account pages as GXP1200 does not have BLF keys
• Added "LCD Backlight Always On" and "Disable Missed Call Backlight" to GXP1200 Basic Settings page
• Changed GXP20xx speed dial key modes and GXP2000 EXT key modes from "Asterisk BLF" to "Busy Lamp Field (BLF)"
• Fixed BT200 redial problem after deleting called log
• Fixed GXP1200 does not reject 3rd incoming call correctly leading to display problems
• Fixed GXP1200 web page problem after 3 concurrent calls
• Fixed GXP1200 busy call forward not working
• Fixed GXP1200/2010/2020 setting DHCP mode in GUI MENU when already in DHCP mode causes it to toggle to static IP mode
• Fixed GXP2010/GXP2020 provisioning screen display not aligned/missing in 1.1.5.6
• Implemented HEADSET key logic so it works similar to SPEAKER key
• Improved handset duplex performance


Post your experiences here:

  • Major : Provisioning using a full template file (as supplied by Grandstream) sends the phone into a constant reboot cycle until the provisioning file is removed, removing P342 (Display Language) as reported below (and a number of similar values) does not fix. The provisioning file does not crash earlier firmware versions such as 1.1.2.25 - Grandstream Ticket 2008020799000203. - Rob
    • Grandstream support indicated that this issue has been fixed in firmware versions 1.1.6.7+. As a temporary workaround they suggested removing the 'P' values for multipurpose keys 8-18 in the template, which worked for me. They further recommended only specifying the 'P' values you wish to change from factory defaults in the template file. - charkins
  • Major : BLF does not work with Asterisk 1.07 from Debian Sarge. Every version past 1.1.1.14 does not work. This is annoying, since I cannot upgrade every Asterisk server I manage. - Kurgan
  • Major : BLF is still not working with asterisk 1.4. In other words BLF is not working on asterisk 1.4 for all GXP-2000 firmware. That's really, really annoying. - Dusty
    • Note : No BLF problems here with Asterisk 1.4 with GXP2000 and GXP2020 with this version. I think you have a problem with asterisk (are call-limit and limitonpeers set in your sip.conf?) - francesco_r
      • For francesco_r: Yes I tried those options like 100 other combinations, no way to show a peer unreachable/unregistered. Please contact me at dustyATnibblesDOTit if you can help, I'll share experience if solved - Dusty
        • This is a bug in asterisk, not grandstream: i have opened an issue one month ago in the the asterisk bug tracker: http://bugs.digium.com/view.php?id=11025 - francesco_r
    • Note : BLF has always been working with Asterisk 1.4. You're probably not setting call-limit and limitonpeers. See: http://www.grandstream.com/documents/GXP2000BLFwithAsteriskConfiguration.pdf, bottom of page 2.
  • Major : no dial tone after sip server not responding for long time, if i call the phone asterisk say "internal server error", i must reboot the phone (the changelog of release 1.1.5.6 report that was resolved but i still have this problem with both gxp2000 and gxp2020) - francesco_r
  • Major : Still unabled to change the GXP2020 Gateway IP if the phone is set with a static ip address. This was supposedly fixed in 1.1.5.6. The address still revert back to 192.168.255.0 even if I set the IP in the config file - Matt Gorecki
  • Major : Phone keeps rebooting after getting provision file if option P342 (Display language) is set. This problem is seen on all version from at least 1.1.4.x. - Bad2Dbone
  • Minor : The italian language pack is wrong, it's german! - francesco_r
  • Minor : Be nice if TRNF key could be edited. - ymartin
  • Minor : Just my gxp2000 was rebooting at least twice a day on this firmware seemingly at random. Never had this problem before and all of our other phones are on the same firmware with no problems. Tried factory reset but did not help. I solved the problem by disabling the watchdog timer. My phone has now been up for 4 days with no problems. I spoke too soon, after 4 days wigged out (all lights strobing very quickly) during a call (ulaw).. Went back to 1.1.4.22 -p
  • Minor : on GXP-2020 the Dialed calls display 24 times the last call !!- furio
    • Note: I cannot reproduce this issue, my 2020 shows dialed calls correctly - Kurgan


Firmware Notes 1.1.5.6 (Unreleased Beta):

Release Notes:

Build 1.1.5.6 (10/22/2007)
• Slow key response fix
• Added GXP1200 LCD Contrast Control
• After Onhook dial the # key and default 4 seconds send number feature cannot work
• keep the PPPOE setting cause the static ip mode setting fail
• Periodic NTP Re-sync not working properly
• 3-way-conference under Notrl MCS mode is not working
• Added syslog if we are unable to allocate anymore channels
• Fix if the server in DNS form is configured, the provision will start two times
• Added "LCD contrast" into the language string
• Updated the new bootloader application status messages to show tftp download and writing messages.
• Implemented vertical status line for gxp1200 for three line display when offhook + softkey.
• For gxp2020 and gxp2010 - cannot change Gateway IP address in static ip mode
• Ported RFC call pickup to be enabled in all GXP
• Changed "Asterisk BLF" string to "BLF"
• Added new menu to reset VLAN under "config"
• GXP2020 offhook screen clears the whole main display area
• Fixed phonebook does not Re-Download atthe specified interval
• BT200: Mute ringer in headset does not work in 1.1.4.20
• Fixed turn off speaker remote disconnect feature for BT200.
• Added two menus to set VLAN tag and priority from the phone
• Fixed: Check SIP User ID for incoming INVITE" checks To header instead of Request URI
• Devices behave incorrectly while in call on receiving NOTIFY (Event: check sync)
• Fixed an issue when the call is on hold, the SOFTKEY is still in OFFHOOK mode.
• Added LCD contrast menu under preference for gxp2020 and gxp2010.
• LCD contrast setting will cause no change if it is not exited properly.
• Added ok and cancel SOFTKEY support for LCD contrast menu.
• GXP2000 1.1.5.3, no dial tone or LED on line key, after the SIP server not responding.
• Presence/BLF button also function as SPEEDDIAL button when idle
• 200 OK for video 'm' line contains an extra space character at the end of the line
• Fixed gxp1200 LED flicker issue



Build 1.1.5.3 (Beta):

Release Notes:

Build 1.1.5.3 (09/20/2007)
• Audio issue on GXP2020
• Added on-hook dialing
• Decoupled registration and subscription intervals
• Fixed we do not re-transmit 200 OK for reINVITE
• Fixed call history time not adjusting to DST
• Support dialing from phonebook with SEND key
• Support for disable backlight when there is missed calls
• Support for BLF call pickup using dialog automatically if XML contains sufficient information
• Support XML application for GXP2020
• Enhanced GUI including reversed MENU, larger fonts for GXP2020
• Integrate my new HLC code + handset gain tricks to reduce ambient noise picked up by handset and eliminate clipping of loud audio
• Remove extraneous lock around packet dequeue loop
• To stop network flood from preventing call setup (tx path works fine in this condition, rx path stutters due to current ne2000 receive buffer overflow handling).


Post your experiences here:

  • Added page up/page down buttons in menus
  • Increased size of all screen elements (less whitespace)
  • New option in 'Basic Settings': 'Disable Missed Call Backlight:'
  • Phonebook Alpha Scroll: Push the corresponding number on the keypad and it will jump to that letter in the list. i.e. press 6 twice to get to the N's in the phonebook! - Shane Steinbeck
  • Audio Quality: I'd say the handset sounds about as good as it has in the last year or so. I think the speakerphone still sucks...I've pretty much given up on that ever working. I bought Aastra 57i's and the speakerphone is flawless. Did I mention I bricked a GXP-2000 and lost more in a flood? Only 4 GXP's remaining :-) - Shane Steinbeck
  • NOTE: Audio quality is improved, I can hear everyone clear with using G729 codec (mnam)
  • On-hook dialing - bcheath

Major : (10/11/2007) On one phone the display is all wrong, the main page is a bit wrong, but the other pages are unreadable. - Richard
    • Note : I can confirm the display problems on my GXP-2000 with build 1.1.5.3, HV: 0.3. I was able to downgrade to version 1.1.4.25 - Knut-Helge
    • Note : I could not confirm any display problems.National characters shows correctly, I have HV:0.4 - Mnam

Major : (10/13/2007) Phone hang ups sometimes without any reason. (ping accessible,but no web + no response from display) - mnam

Major : (10/23/2007) Phone was locked up when I arrived in the office this morning. rebooted. Phone started booting on its own several times later on. It took several minutes to register. Finally it would not register at all. I reinstalled FW 1.1.4.18 - it seems back to normal. (HW=0.4) - - notoncloudnine

Major : (10/24/2007) DTMF RFC2833 stop to works after a transfer. I must set asterisk to INBAND to make DTMF work reliable. The other previous versions worked well. (HW=1.2B) - - francesco_r

Major : (10/24/2007) BLF still unreliable. Sometimes the led is blinking or red when an extension is not in use, and asterisk show correctly the hint idle (HW=1.2B) - - francesco_r

Minor : (10/25/2007) "No Key Entry Timeout" stop to works after the first Onhook dialing (HW=1.2B) - - francesco_r


Firmware Notes 1.1.4.22 (Beta):

Release Notes:

Build 1.1.4.22 (08/31/2007) ----

  • Fixed RTP fails when VLAN is used
  • Handset improvement + packet handler optimization
  • Add Syslog message for off-hook/on-hook notification
  • Fixed we incorrectly drop URI parameters in Refer-To header
  • Fixed GXP2020 one-way audio problem
  • Fixed we incorrectly turns on media when received 18x from caller
  • Fixed we incorrectly turns on media when received INVITE with Replaces and the call its replacing is still in RINGING state
  • Fixed when configured to use STUN and no server is configured we would be re-registering every 30 seconds
  • Added support for early attended transfer (A talk to B, A put B on hold and call C, while C is running A transfer to B) ----
  • Added a "*" suffix to User-Agent field if provider lock is enabled
  • Added audio soft-clipping in TX path
  • Fixed GXP2020 MENU problem reported (Upgrade menu)
  • Fixed BT200 crash on off-hook reported
  • Fixed GXP SIP passwords not stored correctly on additional accounts
  • CN packet handler and generation
  • Fixed random crash problem found in 1.1.4.x ----
  • Changed all GXP to use gxp.lpf to share multi language file
  • Fixed custom idle screen some BMP displayed reversed color
  • Fixed GXP display incorrect EXT key event in factory diagnostic mode
  • Fixed GXP cannot use speed dial key as line keys when all line keys are in use
  • Fixed we turn off speaker on remote disconnect regardless of the setting
  • Added option to change SIP Transport in MENU
  • Added option to configure SIP settings for additional accounts on GXP
  • Added validation check for subnet mask, we will ignore invalid subnet mask
  • Fixed iLBC, G.722, and G.729 problems
  • Fixed out-of-order packets when calling Asterisk voicemail
  • Eliminated the idle screen crash and occasional boot up crash I was still seeing (on 1.1.4.17/1.1.4.18)
  • Fixed AGC does not update TX digital gain, enabling speaker phone to work at longer ranges

  • Louder maximum volume for AER
  • Fixed when a value of 256 or higher is specified in one of the octets for subnet mask, it will be treated as 255. If anyone octet is invalid, the entire address is ignored. This is applicable to Preferred DNS server, IP address, default gateway, DNS Server 1, and DNS Server 2 fields.
  • Changed if invalid (outside 96-127) iLBC payload type is specified we ignore the invalid value (we use to set it to 98)
  • Support for provider lock option
  • Added BT200 display calling number for offhook auto dial
  • Changed Upgrade MENU so changes are saved immediately when OK is pressed instead of needing to choose another Save option
  • Fixed headset volume issue


  • Major : (Oct30/09) BLF is not working with asterisk 1.0.7 from Debian Sarge (old stable). - Kurgan
  • Major : (Sept24/09) BLF is not working with asterisk 1.4.7+ - Dusty
  • Minor : (Jul25/07) DND and BLF broken If DND is activated on a phone using the MUTE/DEL button and the phone is phoned from any another extension also GXP2000, the BLF corresponding to the extension dialed keeps on flashing RED on all phones including the one dialed even after going on-hook. - theobresler
  • Major : (Sep21/07) __Random Lockup. We've had one lockup in 2 weeks and keep the phone sending data to syslog. No message in the log and had to cold boot the phone. Better than previous firmwares, still not perfect. - InDesign Firm
    • Note : (Sept21/07) I can confirm the Random Lockup. I am seeing intermittent (twice in as many weeks) lockups on one GXP2000 that had been perfectly reliable running 1.1.1.4. Oddly, only one of our three phones ever "froze" (requiring a cold reboot / power cycle to restore operation) with 1.1.4.18. The same phone is now locking up under 1.1.4.22 (I never installed 1.1.4.20). The phone that's locking up is the newest hardware of the three and is very seldom used. For this reason, it seems as though this lockup symptom may somehow be related to hardware or the usage (or lack thereof) pattern.- engineerdan
  • Minor : (Sep/07) G.722 problems GXP-2000 using G.722 still has audio glitches. Transmitted audio (from phone) drops out or mutes. Tested with Asterisk and Polycom G.722 - andrew
  • Minor : (Oct/07) BLF does not function when Send Anonymous set to YES under Account - justind





Firmware Notes 1.1.4.20:


This firmware can be downloaded from http://www.bobbyspad.com/

Bug Fixes:



  • Minor : (Jul27/07) Custom Screen Reversed We have built a logo and custom screen for one of our customers. Loaded the firmware and now the screen is reversed.. - InDesignFirm



Firmware Notes 1.1.4.18:

This firmware can be downloaded from the official firmware page http://www.grandstream.com/firmware.html

Build 1.1.4.18 (07/17/2007)
  • Note: Firmware known to lockup on boot if SIP set to use TCP instead of UDP - see notes below

Bug Fixes:

  • Fixed BT200 does not return to idle screen upon attended transfer complete
  • Fixed GXP line number display bug when line number >= 10
  • Fixed GXP2020 WEB UI displays incorrect default SIP port number for Account 5/6
  • Fixed choppy GSM audio quality
  • Fixed G729/G723 may crash in-call

  • Major : (Oct30/07) BLF is not working with asterisk 1.0.7 from Debian Sarge (old stable). - Kurgan
  • Major : (Sept24/09) BLF is not working with asterisk 1.4.7+ - Dusty
  • Major : (AUG17/07) Firmware Release Problems So in 2 weeks worth of conversations with Grandstream Technical support, I finally had them tell me that 4.18 has many known problems and that they have sent it back to their development team for changes. According to the rep I spoke with on the phone, they do not have any new available beta firmware, as they are trying to fix tooo many big problems... I was also provided with a newer version of the firmware for my GXW4104, but was warned that if I installed it, and it didn't work, I couldn't downgrade. Good to see I'm being presented with so many options. On a side note, purchased an Aastra 57i and a Cisco phone. Their is no comparison on how much better their sound quality is. They are absolutely the shizzle. I guess you get what you pay for. - InDesign Firm
  • Major : (Jul31/07) Lockups & Bad Sound Quality We've upgraded several of our GXP2000's only to find that we had a horrible echo when using it with the GXW4104 gateway. We've rolled back to 17 still have it, so rolled back to 4.14 and it has gone away. We've also noticed that in 18 if you set the packets from UDP to TCP the phone just locks up on the next reboot and won't let you get in at all... Had to disconnect from network, reboot, and try to type in the MAC address before it locked up again to reset to factory, just to be able to reload new firmware... I think I'm gonna try an Aastra phone! This is frustrating... - InDesign Firm
    • Note :(Aug20/07) SIP TCP Lockups Configuring to use SIP over TCP will cause the phone to lockup/hang while trying to connect over SIP. DO NOT change SIP over TCP. InDesign's instructions for fixing a locked phone follows. - Tim Smith
      • Note :(Aug23/07) SIP TCP Lockups Unlocked, KINDA The way we were able to recover the phone was to unplug the network cable. Reboot the phone, and then use the on screen menu's to reset to factory settings. At that point, plug back into network, get address, program. If we left it plugged into the network, the second it tried to register, the phone would just freeze. We were never able to get it to work with TCP. You really shouldn't use with TCP unless you are having routing issues, but we were just trying all buttons and options when we found it killed the phone. - InDesign
    • Note :(Aug2/07) Lockups We have 5 GXP2000's, upgraded over the past week to firmware 18, and experience random lockups when idle. Some of them locked after installing the upgrade (from 14), all of them have locked at different times of day - sometimes during office hours, sometimes middle of the night, but only ever one at a time. When they lock the screen appears as normal, the only clue to the lockup is the clock showing the wrong time, and, of course, unresponsive to any keys or incoming calls. Solution is to pull the power and reboot. - olly
    • Note :(Aug5/07) Bad Sound Quality We've upgraded 3 phones to 4.18 and have the same echo problems with GXW4104. I tried all different echo and TX silence settings to no avail. I'm sick of battling and rolled back to 4.14 as well. I must admit that, for future phones, I am also definitely removing Grandstream from my list of possible phones or FXO gateways. - theobresler
  • Minor : (Jul27/07) Ringtones distorted I've made custom ringtones that are bits and pieces of songs. With this firmware, they sound distorted. They sound much better in previous firmwares. (is this because the speakerphone echo canceller?) (HW=0.4).- Joe M
  • Minor : (Jul26/07) Ringtone volume is much louder now. Even at the minimum level it is way too loud when you're sitting just a feet away from it. It's disturbing to others in the same office too. - forsen
  • Minor : (Jul25/07) If DND is activated on a phone using the MUTE/DEL button and the phone is phoned from another extension also GXP2000, the BLF corresponding to the extension dialed keeps on flashing even after going on-hook. - theobresler
  • Major : (Jul24/07) Phone disconnects after 5 minutes after Mute has been actived (HW 0.4). Use case: Listening to a presentation on speakerphone through a conferencing provider. Not sure if the provider disconnects because of 5 minutes of absolute silence from my side? - kam - Disable VAD on your phone. Mute+VAD=No RTP. Most VoIP servers will disconnect because of lack of RTP. - andrew
  • Major : (Jul23/07) Phone freezes in call Using PCMU, HW 0.4 - ninthclowd
    • Note : (July24/07) I have not seen this problem at all on 30 phones. How do you reproduce it? (HW=0.4).- anthony
      • Note : (July26/07) Anthony, as I said, it "happens in call" which indicates that it "randomly happens in call without any sort of catalyst". If you are a GS tech then give me a call (Bill Zhang has my number) and I will get you my config files and screen bg image - ninthclowd
    • Note : I can confirm that the GXP2000 does freeze while in a call using G729, HW 0.2 - bjefferys
  • Major : (Jul26/07) Phone freezes while idle Using PCMU, HW 0.4. I never had this happen before in 1.1.14.17 but it just happened today- ninthclowd
  • Major : (Jul20/07) Breaks BLF light on GXP2000 BLF light is blinking after a call for up to 30 minutes.
    • I can confirm this, i had the problem with both GXP2000 and GXP2020 - francesco_r
    • I have the same problem with the GXP2000 and Asterisk BLF. Other than this the phones rock. -p
  • Major : (Jul23/07) Speakerphone still useless Using PCMU, HW 0.4. various buzzing and volume changes during call - ninthclowd
    • Note : (July24/07) My experience is that speakerphone is vastly improved over 1.1.3.X series, , and I am quite happy with it. You must make sure that you dont mix f/w versions among your phones, as this screws up speakerphone. (HW=0.4).- anthony
      • Note : (July26/07) Anthony, phones should connect to a server and then to another phone in most environments, not by direct IP. This being the case, feedback,buzzing, hissing, etc. are present when doing an echo test to the asterisk server, but only when on speakerphone. This indicates a problem with the phone or the server. In my installation I have many different makes and models dialing outbound through my server with none of these feed back problems. This indicates that I do not have a problem with my server, which makes sense since I am running the latest stable distro of asterisk/zaptel/etc and have tuned my gains to the local telco test line. So the problem is with this particular phone sitting on my desk. Unlike you, I have no desire to move my entire user-base to a beta firmware(that you can't downgrade from) just to fix my speakerphone issues. In the future I would appreciate you keeping your opinion to yourself instead of policing my opinion of whether or not there is something wrong with my phone, newb - ninthclowd
        • Note : (August12/07) After upgrading to .18 from very old firmware that came preinstalled on phone, the speakerphone is much improved. Previously, callee could not hear, while caller could. Now, I have had several successful calls using the speakerphone. And, touch wood, no other problems. However, my setup is very simple, just direct connection to four VOIP providers. - rmclaren
  • Minor : (Jul23/07) Phone volume changes randomly Using PCMU, HW 0.4. While in call, person on the recieving end will hear my voice get spontaneously louder and softer. I believe it's from the new echo canceler
- ninthclowd
    • Note : (July26/07) G726 occasionally sounds quite harsh. This may be the same issue. Not sure about the other codecs. - chewi
  • Minor : (Jul22/07) Some language translations are not viewing menu properly (missing characters in menu).
    • Slovak language is affected. - mnam
  • Note : (July20/07) Receptionist (with extension unit) at 75 phone location reports this is "Perfect".- diver
  • Minor : (Aug10/07) My call history time stamps are off by -1 hour. The time display is set to the correct time. Anybody else seeing this? - Shane Steinbeck
    • Note : (Aug13/07) yup, mine too - olly
  • Note : (Aug13/07) It would be handy if the status page showed the time/date on the phone so that these settings could be checked via the Web interface without having to visit the phone or asking someone to look.- Linker3000
  • Minor : (Aug13/07) The LCD backlight turns itself off during calls - I'm sure that in previous version, the light was illuminated all through the call. - olly
  • Minor : (Sep 4/07) Time of call does not include DST (daylight-saving time) - When you check call history, the time of call is displayed wrong. - bad2Dbone
  • Minor : (Sep 4/07) Phone keeps restarting with language pack if you are using provisioning with preseted language (tested with german) - bad2Dbone
  • Minor : (Sep 4/07) Noise in headset is much lower than in previous version, but it's still there. (professional headset with amplifier) - bad2Dbone
    • Note : (11.9.07) Noise appears sometimes after hanging up. It lasts for a while- bad2Dbone
  • Major : (11.9.07) Transfer sometimes doesn't work A calls B. B answers, picks new line and calls C. Than B picks back line with A, presses TRNF and line vith C. C hears silence, and cann't be picked up again - bad2Dbone







Firmware Notes 1.1.4.17:

This firmware can be downloaded from http://www.grandstreamsucks.com or http://www.idlex.net/?p=10

Build 1.1.4.17 (07/13/2007)

BugFixes:

  • Fixed LCD backlight switched off at inconsistent interval after off hook
  • Fixed GXP2000/2020 do not prompt user in dialing screen if account is not registered
  • Fixed incoming direct IP calling will not work if no accounts are configured
  • Fixed GXP2000/BT200 handset overdriving in TX path
New Features/Changes:
  • Changed we will place the phone on-hook after transfer is completed

  • Major : (Jul20/07) Breaks BLF light on GXP2000 BLF light is blinking after a call for up to 30 minutes.



Firmware Notes 1.1.4.16:


Build 1.1.4.16 (07/06/2007)

BugFixes:

  • Fixed GXP2000 noise on call hang up
  • Fixed GXP does not allow change network settings when "Lock Keypad Update" is set to Yes
  • Fixed GXP2020 screen does not clear cleanly when going offhook (some pixels left over from the date string at idle screen)
  • Fixed GXP2020 does not show SRTP icon when making an SRTP call
  • Fixed GXP2000 increased handset background noise in 1.1.4.14
  • Fixed BT200 does not play ring tone after call-waiting came in and onhook
  • Added BT200 onhook call-on-hold reminder message (port from BT100)
  • Fixed BT200 HOLD issue: hanging up the handset while having a call on HOLD will terminate the call
  • Adjusted GXP2020 GUI line info display for multiple call scenarios
  • Fixed handset ringer level too loud for GXP2000
  • Disabled silence suppression for G723/G729 to bypass the crash the problem
  • Fixed we will default to PCMU regardless of configured codec when incoming SDP contains video
  • Added option to disable use of multiple media attribute in SDP to workaround some platforms not supporting it. Provisioning parameters P137/487/587/687/1787/1887, takes immediate effect without reboot.
  • Fixed GXP2020 does not show "Sync Phonebook XMLâ?¦"
  • Fixed we use ";user=phone" in BLF SUBSCRIBEs even when "User is phone" is not selected
  • Fixed we will only update the first BLF/Presence status
  • Fixed GXP2000 high pitch sound when offhook using the handset
  • Fixed we do not honor session-timer refresher party when other the parameter contains a space between semicolon and "refresher=" parameter
  • Added support for Nortel MCS server-side conference
  • Added provisioning status on BT200
  • Added GXP offhook auto dial
  • Added support for BT200 provisioning display (Bugzilla #748)
  • Added support for new factory parameter section (to be removed for external release note)

  • Major : (Jul11/07) Line remains off hook after transfer After successfully transferring a call and hanging up, line1 remains illuminated red and off the hook. (HV .4). This is a real annoying bug. - JoeM <- FIXED IN 1.1.4.17 (iDLEx)

Firmware Notes 1.1.4.14:

Although not available as an official download as of Jun26/07, Grandstream support is mailing this firmware to people who request it.
This firmware can be downloaded from http://www.grandstreamsucks.com/ No pun intended!
Changelog 1.1.4.14:

Bug Fixes:

  • Fixed SRTP broken in 1.1.2.1
  • Fixed SRTP sequence number wrap around bug
  • Fixed we include "application/xpidf+xml" in Accept header for SUBSCRIBE
  • Fixed GXP-2000 Accounts 1 and 4 "SIP T1 Timeout" last option incorrectly labeled as "1 sec" instead of "2 sec"
  • Fixed we stuck at provisioning when receives a 200 OK with 0 content-length
  • Fixed DoS issue by WWW-Authenticate header
  • Fixed a crash issue with 489 Bad Request caused by Proxy-Authenticate header
  • Fixed we cannot parse DST string via configuration file correctly
  • Fixed ring tone download goes to incorrect TFTP server
  • Fixed we still claim an IP address after receiving DHCP NAK
  • Fixed description for "Enable Call Features" in web UI
  • Fixed we incorrectly formed the qop param without the quotation
  • Fixed we incorrectly parsed the nonce param
  • Fixed we incorrectly handled presence NOTIFY
  • Fixed we incorrectly handled dialog NOTIFY
  • Fixed we incorrectly responds 481 to refer NOTIFY when BYE arrives first
  • Fixed we accept broadcast SIP messages
  • Fixed phonebook download account index incorrect
  • Fixed GXP-2000 crash with EXT board
  • Fixed GXP-2020 always sends signal=2 in DTMF via SIP INFO regardless of the actual DTMF
  • Fixed the in-call timer does not tick when a call is on MUTE
  • Fixed the offhook status line disappears between DTMF digits and in-call
  • Turn MWI off when phone is offhook
  • Fixed phonebook download account index incorrect
  • Fixed CBCOM mode results in no audio
  • Fixed we display "HV: 0.4.255" on some old GXP-2000 hardware
  • Fixed Sylantro interop issue
  • Fixed we do not honor maddr parameter in SIP Contact header
  • Fixed we used the cached "realm", "nonce", or "opaque" parameter if they are 0 length
  • Fixed we do not display correct IP address after DHCP NAK
  • Fixed GXP2000 crash if you add entry to phonebook from call log and edit name/number field
  • Fixed we cannot authenticate using auth-int
  • Fixed a TCP interface bug and a HTTP server bug which should speed up web UI and fixes some display issues

New Features/Changes:

  • Improved audio quality
  • Changed description for "Enable Call Features" in web UI
  • Changed we use 408 instead of 487 for ringing-no-answer
  • Added wrap-around support for MENU UI
  • Added speed searching phonebook and other list box items
  • Added option to support mute speaker ring in headset mode (P336) --Note that this option does not exist for GXP-2000 HW 1.0
  • Added we will clear new missed calls display message after viewing Missed Calls menu without visiting details of each entry
  • Support for concurrent multiple DTMF schemes
  • Changed: We will bypass unregister and move forward to register if we received a non 2xx final response for the the (un)-REGISTER
  • Added we will clear new missed calls display message after viewing Missed Calls menu without visiting details of each entry
  • In web UI, the "if set to Yes, "#" will be function as the "(Re-)Dial" Key" is removed
  • Add display "Preparing to write" and "This may take a while" between the period of firmware download complete and firmware flashing begins
  • Disabled iLBC (iLBC is not working well and will be fixed later)
  • Added support for XML encoded in UTF-16
  • New dialing string display-old scheme will only show the last 11 digits dialed; new scheme will automatically scale down to a smaller font to display up to the last 42 dialed digits. Old scheme does not show line and account info while dialing digits, new scheme will.
  • New centralized GUI control allowing showing multiple call information simultaneously including:
  • Still show account information while dialing (previous version only show dialed digits)
  • When 2 calls are present, we will split screen in half vertically (3 lines per call) and display both calls
  • When 3 calls are present, we will split screen in 3 sections vertically (2 lines per call)
  • When 4 or more calls are present, we will split screen in multiple sections (1 line per call) and display as many calls as we can. If there is an active call, that call will occupy 2 lines.
  • Select scenarios will occupy the entire screen for the active call (such as when TRANSFER key is pressed, or when CONFERENCE key is pressed)
  • When TRANSFER key is pressed you will still see the current calling info along with the prompt or the transfer target number you are dialing
  • When CONFERENCE key is pressed you will still see the current calling info along with the prompt
  • When call forward are requested (such as *73), you will still see the account information along with a prompt and the forward target number as you enter

  • Upgrading to this firmware can take several reboots - be patient and wait for them to all be completed before logging back in to your phone/s.

  • Major : (Jun30/07) Incoming call on second line - no voice after pickup call (headset or speaker), first line is OK Incoming call on first line worked well, the only way hot to enable sound on second line afte pickup call is to hold the line and resume call, before that there is no sound on both calling parties . - Pteppic

  • Major : (Jun27/07) Speaker always on Speaker hisses even if no call active (HW v2.0). - Tigs

  • Major : (Jun27/07) Line remains off hook after transfer After successfully transfering a call and hanging up, line1 remains illuminated red and off the hook. This is unacceptable! - JoeM
  • Note : (Jun28/07) confirmed. HV=0.4 - ampster

  • Major : (Jun26/07) Breaks HOLD button on BT200 It has been reported in the Trixbox forums that this firmware stops the HOLD button from working on the BT200 model. GXP-2000 not affected. - Linker3000

  • Question : (Jun26/07) I'd like to know, whether the new firmware policy will be to issue upgrades only upon request, whether this firmware is considered stable or not (I don't get the point in the question marks behind the word stable... shall this be considered 'beta, although (of course) it's hoped to be stable'?). and whether there will be new official stable images distributed via the official site (as opposed to 'only upon request')? - kodomo
    • Answer: (Jun29/07) One of the engineers at Grandstream contacted me and asked me to post this response to some of the questions about the firmware and these pre-beta versions we keep seeing. I think he addressed this situation very well, and I will summarize by saying: "Of course it's broken and there are bugs with it. It's pre-beta. Grandstream appreciates all of the feedback on problems, but it does not make sense to expect the pre-beta firmware to be stable or even have all of the functionality one would expect from a stable release or full beta release candidate." Upgrading is a one way street, and they warn time and again against doing it in production environments before the firmware is released as stable. Anyway...back to the engineer's response: - thetatag
      • We did a lot of hard work to get to 1.1.4.14, especially with the audio fix. That is not an easy one given that we cannot change the hardware case to maximize the physical sound path. Initial internally rolling testing seemed fine, but we did find some test phones needed some time to get stabilized (they will freeze after f/w upgrade) and the chirp sound in on/off the handset. Also the BT200 HOLD key is broken (although the FLASH key can do a similar thing as HOLD). That is why we halted it in pre-beta and did not even go to beta as we try to nail down these issues before publishing into beta, because the update is a one way train.

  • Note : (Jun28/07) Yes, I have some background hissing (white noise) but the firmware is so much better than any before. THANKS.- richard
  • Note : (Jun28/07) I have always had some slight but barely noticeable hiss (white noise). HV=0.4 - ampster

  • Note : (Jun28/07) some improvement in speaker phone also. but there is an initial burst in speaker mode. HV=0.4 - ampster

  • Note : (Jun28/07) Overall, the firmware keeps on improving. HV=0.4 - ampster
  • Note : (July04/07) First version in a year or more that DOESN'T BLANK THE SCREEN (HV0.3). That is very good, it has worked very well for 2 days since installing the new FW. Less good though - POSSIBLE WARNING - My phone died today, it is now a flashing brick. I can't rule out that it has to do with this new firmware that has been on for 2 days.- carlotto



Firmware Notes ("Beta" 1.1.1.13):


Changelog (1.1.1.13)
  • Audio adjustments
  • Fixed some v 0.3 and 0.4 blank LCD issue
  • Added "application/dialog-info+xml" in the Accept header of 415 response
  • Added Syslog for SIP dialog matching result
  • Fixed init problem causing the GXP-2000 v1.1 bootup problem

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.1.13.pdf" file included with the firmware. - flu

Bugs / Tweaks (1.1.1.13):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR: (Sep29/06) SCREEN BLANKING Took about 15 mins for it to blank. Come on Grandstream, this is getting very very tiresome. Carried over from the mists of ancient time, onwards. - mattb
    • NOTE: (Oct01/06) Hey mattb, have you tired another power adaptor? I had one of my phones doing crazy things and found that the power adaptor was giving out strange voltages, changed out and works like a dream. - jase495

  • MINOR: (Oct02/06) GUI still is very ugly and hard to use. Please consider the changes your users suggested below (search for "low usability" on this page). - job

  • MINOR: (Oct02/06) The naming of line 10 and 11 (":" and ";", respectively) is unexpected enough to be called a bug. If it's impossible to call them "10" and "11", go for "A" and "B" or something. - job

  • MINOR: (Mar29/06) Numbers off-screen. When entering long phone numbers (i.e. for international calls) the phone number goes off-screen as it advances, rather than moving up to the upper line as it used to. Carried over from 1.1.0.13 onwards. - Mike

  • MINOR: (Sep24/06) State Bugs. Loopback call (dial phone from itself, via asterisk) drops out on answer (as expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up/down arrow adjusts the volume instead of going to phonebook. Carried over from 1.1.0.13 onwards. -Jedi98

  • MINOR: (Oct05/06) Call timer on LCD appears to wrap around, presumably at 12 hours. Noticed this while testing phone stability. - edgar

  • Question: (Sep30/06) Audio quality. Is the audio any better in this release? I'm still on 1.1.0.11 and dont feel like risking it unless audio is OK. - falle
    • A: (Oct01/06) IMO Audio was OK upto and inc. 1.1.1.7 and then wen muffled. At 1.1.1.13, g711 (PCM) is better than 1.1.1.12 but not perfect. g729 remains muffled by comarison to 1.1.1.7. Other users have had different audio issues but I am not sure if these have now been solved. I would advise holding off if possible, unless you have other bugs you need fixed, since you cannot downgrade. .13 is too new for all the bugs to be in yet. - jedi98
    • A: (Oct01/06) Yes, I'd say it's "different". It's much better than before, but the sound was clearer in 1.1.0. - job

  • Note : (Oct05/06) Stability. I have found that 1.1.1.13 seems to have fixed stability issues I saw with some of my phones on 1.1.1.9. These phones would often hang at boot and sometimes freeze both on and off calls. So far, these upgraded phones have not frozen on/off calls or hung at boot. I will probably be upgrading all my phones to 1.1.13 from 1.1.1.9 after testing it on a few user desktops for a while. - edgar




Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)

  • MAJOR: (Oct 2/06) Sidecar stopped working in v1.1.1.13 (was working with v1.1.1.12), red light just flashes fast (uninitialized). Not using AC adapter to power sidecar. - andrew4455
    • NOTE: (Oct03/06) Confirmed, Red flashing light, does not initialize. Rebooting from GUI crashes the phone. - joekane


  • MINOR: (Oct 2/06) I called tech support about the audio quality when using the sidecar, and was told to NOT use the ac adapters with the sidecar(s) with the GS 2000 hardware version at 1.1. I took off the adapters, and the audio quality is better, but not quite good enough. I am on .12, and waiting to see if there are other issues with .13. Current problem is that voice is choppy at times.




Firmware Notes ("Beta" 1.1.1.12):


(Sep22/06): Currently available from Grandstream BETATEST site, 1.1.1.12 Firmware, 1.1.1.12 Release Notes.

Changelog (1.1.1.12)
  • No change for GXP-2000 from 1.1.1.11
  • Disable the EMIF optimization which caused some BT-200 to fail to upgrade


Bugs / Tweaks (1.1.1.12):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR (Sep24/06) Audio quality on speaker and handset is still muffled (by comparison to 1.1.1.7). Codecs tested were 711 & 729. High frequencies are cut and distorted, it sounds like the phone has a slightly stuffed up nose (A phone virus maybe? ;)) Carried over from 1.1.1.9 onwards. - jedi98

  • MAJOR (Sep25/06) Unable to Register via LAN to LAN. Example, the phone is on my private lan (192.168.11.0/24) and my default gateway has a VPN session to my office LAN (192.168.44.0/24). The Grandstream GXP-2000 registers fine with my local asterisk server (192.168.11.10) but won't register with my office server (192.168.44.2). Checking tcpdump/stats the phone doesn't even send the sip to the default gateway, perhaps its not sending to d/g because it sees the IP as in RFC1918 and is hard-coded in the firmware that all rfc1918 doesn't need to go via the gateway? (just a guess!) Bug is present in ALL previous firmware releases tested, at least back to 1.1.0.16 - andyb2000
    • NOTE: (Sep25/06) I have similar config but I am NOT getting the problem. I register acct 1 with a local * server and acct 4 with an * server on another subnet in vpn. Subnets are 192.168.1.0/24 and 192.168.11.0/24. VPN is done by routers. So there must be somthing differing in our configs somewhere. - jedi98
    • NOTE: (Sep25/06) Interesting, would you take a look at my profile page here and get in touch? Many thanks - andyb2000

  • MINOR: (Sep24/06) State Bugs. Loopback call (dial phone from itself, via asterisk) drops out on answer (as expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up/down arrow adjusts the volume instead of going to phonebook. Carried over from 1.1.0.13 onwards. -Jedi98

  • MINOR (Sep26/06) Speakerphone mic too sensitive. In 1.1.1.10, they increased the speakerphone mic gain, and now it is much too sensitive. It picks up all the background noise, and it sounds (to the other side) like you are in a train station. Ideally, the gain should be adjustable in config. - bcheath

Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)

  • MAJOR (Sep26/06) Extension Unit hanging In 1.1.1.12, Using asterisk 1.2.7.1-BRIstuffed , Maybe its the asterisk version im using but the extension unit just hangs after about 10 minutes of operation. A reboot fixes it. - joekane

  • MAJOR (Sep26/06) Paging Issue In 1.1.1.12, Using asterisk 1.2.7.1-BRIstuffed , When paging to 40 GXP's alot of the lights still stay on even tho remote disconnect is enabled on all phones. Reboot fixes. (Phones on 1.1.0.16)??? - joekane

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Firmware Notes ("Beta" 1.1.1.11):


NOTE: (Sep16/06) Firmware 1.1.1.11 has been pulled from beta site. Kurgan


NOTE: (Sep14/06) The filename suggests this is a Release rather than a Beta although it appears only on the Betatest site currently. - Channel-Two
  • (Sep14/06) The betas always say Release_GXP2000-BT200_version, I don't understand how this one is different. I think they're only considered stable when they are in the DOWNLOAD directory instead of BETATEST. - Shane Steinbeck
  • (Sep15/06) Okay fair enough. - Channel-Two

Changelog (1.1.1.11)
Fixed Phonebook problem again?

Bugs / Tweaks (1.1.1.11):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR (Sep15/06) Audio quality on speaker and handset is still muffled (by comparison to 1.1.1.7). Codecs tested were 711 & 729. High frequencies are cut and distorted, it sounds like the phone has a slightly stuffed up nose (A phone virus maybe? ;)) Raised to major because audio is the phone's primary function!. - jedi98

  • MAJOR: (Sep15/06) Display still blanks after some time. MAC 00.0B.82.03.XX.XX, H/W v0.3. Accessing the phonebook will cause it to come back, unfortuantly picking up the handset won't. Sometimes triggered by a call. - jedi98
    • NOTE: (Sep15/06) I confirm this, again. BTW, this is my first and last phone from Grandstream. - julianjm
    • NOTE: (Sep15/06) Confirmed - mattb
    • NOTE: (Sep16/06) I have never experienced display blanking, I have about 20 phones, the oldest has MAC 00.0b.82.05.aa.02. It definitely means that it's a hardware issue, probably in some specific HW revisions. Maybe GS should clearly state what HW revisions are faulty, and stop trying (but did they even actually tried, or are we just hoping that the are trying?) to fix this bug in firmware. - Kurgan
    • NOTE: (Sep16/06) The point to remember here is that when these phones came out with 1.0.1.9 firmware there was no problem with the disp. at all. But some functions were to be added with later firmware, so the phones were essentially sold with functions that required new fw to use them. Upto at least 1.0.1.13 there was no problem here, but those who went beyond cannot downgrade and the .03 rev phones were on sale until very recently. - jedi98
    • NOTE: (Sep19/06) Strangely enough i have 3 of my MAC 00.0B.82.03.XX.XX phones running 1.1.1.11 and none of them are having blanking problems (that we have noticed), every firmware that i have tried until now has caused the problems though... - SoloFlyer
    • NOTE: (Sep19/06) Scratch that.. my phone just blanked twice within about 15 mins... - SoloFlyer
    • NOTE: (Sep20/06) Ok, so it seems that newer firmwares trigger the display blanking on phones that previously worked well. I supposed that on older phones any firmware after the legacy 1.0.0.x caused the problem, but I seem to be wrong. - Kurgan

  • MINOR: (Sep15/06) State Bugs. Loopback call (dial phone from itself, via asterisk) drops out on answer (as expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up/down arrow adjusts the volume instead of going to phonebook. -Jedi98

Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)


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Firmware Notes ("Beta" 1.1.1.10):


Changelog (1.1.1.10)

Release Note
Firmware Version 1.1.1.10
September 13, 2006
Firmware 1.1.1.10 has major changes (compare with 1.1.0.x Releases) which may requires firmware to be downloaded twice(and reboot itself), and it can not be downgraded to previous version.
Make sure all the files that come with
Release_GXP2000-BT200_1.1.1.10.zip is unzipped into the TFTP or HTTP server.
For any firmware upgrade from 1.0.1.x or 1.0.2.x, please refer to previous release note and firmware and upgrade them to 1.1.0.16 first.
===============================================================
Product: GXP2000
Date: 2006-09-13
Release items: boot55a.bin 1.1.1.2
gxp2000a.bin 1.1.1.10
Previous release: boot55a.bin 1.1.1.1
gxp2000a.bin 1.1.1.9

Release Note for GXP-2000 and BT200
Build 1.1.1.10 9/12/2006

  • Fixed GXP-2000 crashes when a very large phonebook file is downloaded
  • Fixed GXP-2000 crashes when "Remove Manually-edited entries on Download" is set to Yes
  • Fixed GXP-2000 Name is not displayed for multi-functional keys on EXT
  • Fixed SIP stack incorrectly parsed "CT" header
  • Turn on this option by provision parameter P339 (1: use Account Name, 0: use date). First 12 digits are displayed, aligned to center (odd length 1 slot to the right), undisplayable characters will be blank
  • Support for 5 provision attempts
  • Fixed GXP-2000 take account 1 information when replying the missed call for account 2
  • Fixed GXP-2000 Speaker mode is triggered on 2nd call when the first call ended
  • Fixed audio quality degraded when call-waiting tone is been played
  • Fixed we cannot correctly parse incoming SIP messages with Contact headers that come without a username part causing some Broadsoft test cases to fail
  • Support for Broadsoft click-to-hold (Allow-Events: hold)
  • Fixed G.723 6.3kbps decoder does not work, web UI enabled this option
  • Increased speakerphone mic gain by 7.5db

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.1.10.pdf" file on the BETATEST download site. BT200 Specifics removed - Shane Steinbeck

Bugs / Tweaks (1.1.1.10):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR: (Sep14/06) Speakerphone: Sounds terrible "like you are in the twilight zone." - Shane Steinbeck
  • MAJOR: (Sep14/06) Screen: Screen still blanks after a few minutes... yawn.... - mattb

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Firmware Notes ("Beta" 1.1.1.9):


NOTE: (Aug14/06) NEW file appears to be available http://www.grandstream.com/BETATEST/GXP2000_BT200/Release_GXP2000-BT200_R1.1.1.9.zip though it contains boot55b.bin bt200b.bin gxp2000b.bin and my phone only tries to get the a version of these files. - Andy
NOTE: (Aug14/06) The missing "a" versions have been added by Grandstream today by updating the firmware file. I was able to upgrade to 1.1.1.9. - kam
NOTE: (Aug14/06) I had no problem with upgrade to 1.1.1.9 with "b" on my GXP2000 with 1.1.1.7 previously installed. - kondor

Changelog (1.1.1.9)

  • Improved audio quality
  • Modified memory management for iXML parser. This should resolve the freeze on downloading 50-record phonebook XML problem
  • Fixed several GUI menu bugs Fixed GXP-2000 does not save after more than 30 extension entries
  • Fixed GXP-2000 does not store UserID for KEY36
  • Fixed GXP-2000 cannot answer incoming call when in the SIP proxy edit screen
  • Fixed the screen XML '$d' variable does not display correctly
  • Fixed BLF does not activate speed dial when BLF party is in use
  • Fixed GXP-2000 continue to ring when BYE is received for early dialog
  • Fixed we do not clean out the call properly when terminating a call due to SRTP not enforced
  • Fixed we will perform firmware upgrade even if configured not to when DNS query for config server failed/we query "0.0.0.0" when configured such in firmware/config servers
  • Fixed a memory-leak issue that is only exposed by how GXP-2000 handles attended transfer (does not apply to other products)
  • Fixed GXP-2000 does not place transferee on hold when attempting to transfer
  • Extend the original "Disable missed-calls" feature to allow a new mode to disable all call-logs on a per-account basis. P182/442/542/642: old values (0/1), new values (0/1/2) where 2 means disable call-log.
  • Fixed both GXP-2000 and BT-200 turns speaker on when MSG key is pressed even when no voicemail user ID is configured
  • Fixed a potential crash if a NOTIFY with bad dialog XML
  • Added a memory debug feature: on right-top corner current memory status is displayed in lieu of time (or date, if reversed) in the format of x/y where x is the current usage and y is the peak usage

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.1.9.pdf" file included with the firmware. BT200 specific changes were removed. - flu

Bugs / Tweaks (1.1.1.9):


Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR: (12 Sept 2006) Voice Mail UserID: When this is set, the phone is unstable and locks between 5 and 20 minutes. This value was set to the shortcode for checking VM. If this is incorect usage, the phone should fail gracefully.
    • NOTE: (Sep19/06) I haven't seen this behaviour with about 45 running GXP2000 phones - edgar

  • MAJOR: (Sep05/06) RTP Decode: When the GXP-2000 receives an RTP packet, it assumes that the packet is encoded with the same codec that the GXP is currently sending. This need not be the case. The phone should look at the payload type byte in the packet to decide which decoder to use. The result is one way audio even though the RTP stream can be seen to be flowing in both directions correctly. - marvy

  • MAJOR: (Aug31/06) Call Waiting: When call waiting tone is enabled and a second call comes in, the audio of the call you are on is effectively muted permanently. - forrestc

  • MAJOR: (Aug15/06) Phone Book: When attempting to update phone book and having the option "Remove Manually-edited entries on Download" set to Yes, the phone still freezes (clock stops, doesn't ring on call, totally unresponsive until power cycle) with "Sync Phonebook XML... This may take a minute" on screen. - acabtp

  • MAJOR: (Aug16/06) Lockups: Seems to randomly lock, I'm using BLF so it may be when states of those change, however after about 10mins the phone totally locks up and needs power reset - Andy
    • NOTE: (Aug14/06) You will want check your network for odd behavior. Most of the phone lockups I found were due to bad patch cords or misbehaving equipment on the network. My T1 at one point was the culprit. Due to a bad smart jack the te110p was sending out false data to certian phones. Another cause is the syslog. With the syslog enabled on too many phones our network took a huge hit and many phones acted up. - diver
    • NOTE: (Aug23/06) Not based on cabling, etc as those were already checked, but 24hrs later and a few reboots the problem has gone away, as has the web interface crash below - Andy
    • NOTE: (Aug31/06) I'm also seeing locks where you will be talking on a call and the phone simply locks up - no audio, can't hang up, etc. Needs power cycle to recover. I'm almost 100% sure there are no network issues causing this. - forrestc
    • NOTE: (Sep19/06) Seeing random lockups on a few phones, never while in use. These phones also tend to have problems coming up after reboot. Reflashing doesn't seem to fix so I am RMA'ing phones doing this. - edgar

  • MAJOR: (Sep07/06) Display still blanks after some time. MAC 00.0B.82.03.XX.XX, Pressing certain buttons on the hand set will cause it to come back, unfortuantly picking up the handset wont - SoloFlyer
    • NOTE: (09/08) Happens with MAC 00.0B.82.03.CC.2F. HW revision: 0.3. It seems, that 0.4 is OK - FESTR

  • MAJOR: (Aug16/06) Web Interface: Trying to access web interface, allows password/login but as pages start to load they freeze after perhaps one or two input boxes can be seen, phone then crashes, needs power reset. - Andy
    • NOTE: (Aug23/06) 24hrs later and a few reboots the problem has gone away, as above (lockups) - Andy

  • MAJOR: (Sep/06) Registration: Phones often do not reregister if they lose their connection to asterisk, for example if the server is rebooted. Often they also stop responding to pings and the web interface is inaccessible even if the phone is forced to reregister. I force this by running the ethernet loopback test for a few seconds after which the phone will reregister in about 30 seconds. This is a major inconvenience if you have to go to most of your phones and manually make them reregister! - edgar

  • MINOR (Sep19/06) Cannot access the web interface if phone is in use. You get a message saying the phone is busy. - edgar

  • MINOR (Aug15/06) Audio on speaker and handset is muffled by comparison to 1.1.1.7. Codecs tested were 711 & 729. I downgraded the phone to 1.1.1.7 and the audio was clear again. It sounds like the volume was boosted slightly at 1.1.1.9 but the high frequencies were cut or distorted. - jedi98
    • NOTE: (Aug24/06) I can confirm a quite bad sound (distortet and some freqs cut off). AGC on handset's speaker still gives bad volume changing. - fratzr

  • MINOR (Aug15/06) The audio crackles on g.726 codec (the same on 1.1.1.7 firmware). - Maxo

  • MINOR (Aug23/06) Every few mins using BLF I am getting the following in asterisk logs, and the phone doesn't update the BLF indicator for that SIP channel.
Incoming call: Got SIP response 415 "Unacceptable Content-Type" back from 192.168.2.2
(that IP being the phone that has the BLF indicators set)
- Andy

  • MINOR (Aug03/06) - BLF State Freezes For some reason BLF seems to work fine on my phones for a while, but then something causes all of the lights to freeze in whatever state they were in (i.e. ringing, idle, off hook) and they won't refresh until reboot. I just upgraded to asterisk 1.2.10 around the same time I upgraded my firmware. Prolly a dumb move, however I am getting no errors whatsoever from the asterisk logs. Anyone else having this problem? - ninthclowd
    • NOTE: (Aug08/06) I think I found out what is happening. Lets say there are two phones, A and B. Phone A has a BLF button setup for Phone B. Something then causes phone B to lose its registration until reboot <tangent> I have no idea why this is happening but it only started happening this firmware and should probably be listed as a major bug in itself as it will drop calls </tangent> The BLF state for Phone B on Phone A will freeze in whatever state it was in when it lost registration and won't clear until reboot. - ninthclowd
    • NOTE: (Sep10/06) I am having this problem with 1.1.1.9, but I don't think my problem is being caused by lost registrations. - vinceval

  • TWEAK: (Aug14/06) SIPS support: It's really nice to have SRTP now - however, with SDES the master password is sent unencrypted in the SIP stream AFAIK, so for real security it might be useless... Therefore the GXP-2000 should also support SIPS to make this solid (or is it already supported?) - Mirak

  • TWEAK: (Aug16/06) Phone Book: The numbering system in the phonebook has changed. In the previos version the phone lines numbering started with 0 and end winth 3. In the new firmware the lins are numbered as they are on the front panel, Line 1 is 1, Line 2 is 2, Line 3 is 3 and Line 4 is 4. I've made a XML file phonebook and I had to go +1 on every entry in <accountindex> </accountindex>. - kondor

  • TWEAK (Aug22/06) built in microphone When you use the handset and press the speaker button the microphone should not switch to the build in microphone but should stay in the handset. This function is used for letting others listen to a conversation without notice, when the handset is on hook the built in microphone should be used. Right now its not possible to let someone listen to a conversation without lower quality for the called party, even old ISDN phones have this feature.- Datu
    • AGREE WITH YOU (Sept16/06) This feature is very common and will arrange echo problem...- flo_turc

Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)

  • MAJOR: (Sep11/06) Sidecar: Buttonnames aren't displayed only the telephone number by pressing a speeddail key on the extension unit. The speeddial keys on the GXP-2000 does display the name and telephone number. - LJPYRO

  • MAJOR: (Aug16/06) Sidecar: Still doesnt work with using the web interface, buttons pased 18 dont light up *Asterisk BLF*. this is very disapointing Grandstream. Selling the hardware months before it can work, get it sorted will you - Joe

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Firmware Notes (Beta 1.1.1.7):


NOTE: (Aug11/06) The file has been pulled from the beta download site. - Shane Steinbeck

Currently available from Grandstream. This is also a browsable directory, more stuff available at Grandstream downloads or Grandstream Beta Site Home.

WARNING: This is a BETA version. It may have stability issues and is provided by Grandstream for testing purposes only. Do not use in a production environment untested.

Changelog (only GXP-2000 related)
Build 1.1.1.7

Fixed custom ring tone by Alert-Info fails
Added option to check incoming INVITE sip user ID
Fixed DTMF buffer not cleared when switching lines for unestablished dialogs
Support disable call-waiting tone
Add UCF (Unconditional Call Forward) icon on status line
Fixed high pitch done played when Call Forwards are enabled and disabled
Fixed user cannot enter * and # in phonebook entries. In addition, user can enter @ by using HOLD key in phonebook submenu
Fixed we crash on attended transfer on platforms that use To/From headers without square brackets
Fixed we still responds "recvonly" on un-hold SDP message
Fixed GXP-2000 ring tone change via keypad menu not effective after reboot
Added volume control is stored after reboot
Added Support for GXP-2000EXT keys in diagnostic mode
Disabled headset side tone
Fixed IP Fragmentation bug
Add Support for IM and screen XML feature (saving to flash)
Fixed we send NTP to wrong IP address
Added force LCD update on hook status change (this makes LCD GUI look more responsive when onhook)
Added customizable idle screen via downloading XML by HTTP/TFTP
Added support for SIP MESSAGE method (RFC 3428); stores up to 100 incoming IM messages, after that new messages are dropped
Added support for SIP PUBLISH method (RFC 3903)
Added support for SIP Presence package (RFC 3856, 3863) for use of 7 MFKs and GXP-2000EXT
Added support for SIP Dialog package (RFC 4235)
Added support for SRTP by SDES
Fixed GXP-2000 crashes when speed dial user ID contains '@'
Fixed the clock on the right top corner displays incorrectly if switches from 12hour display to 24hour display.
Added support for G.726 codec
Added support for GXP-2000EXT console.
Added support for anonymous call using privacy header
Added support for downloadable phonebook

  • IMPROVED: Speakerphone volume has been improved. This is a work-in-progress as Grandstream is working hard to preserve accoustic quality while increasing the volume of the speakerphone. It hasn't "arrived" yet, but it's moving in the right direction. - thetatag
  • IMPROVED: Forced screen refreshes on certain events takes care of many of the screen problems with older hardware. - thetatag
  • IMPROVED: Handset, spealerphone, and headset volume are now all independent.- thetatag

Bugs / Tweaks (1.1.1.7):

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(Jul28/06) Could you please describe what a useful comment would contain? thanks. - Anthony

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  • MAJOR WARNING: (Jul28/06) There is no Turning Back I tried going back to firmware 1.1.0.16 through tftp and reboot, no luck. So I tried a full reset using the menu and the MAC address. The phones still have firmware 1.1.1.7 loaded. duh. I was hoping to recover the audio quality of the older firmware, no such luck. - Anthony
    • NOTE: (Jul29/06) Also got caught out. Tried to downgrade with HTTP. But, I'm now so used to getting caught out by these betas that I gave up complaining! Plea to GS: If you cannot provide a method for downgrade then CAN YOU PLEASE show a big warning with the notes for a non-reversable releases (BETA or otherwise). - jedi98
    • NOTE: (Aug08/06) /Agree with the Jedi - ninthclowd

Extension Unit / Sidecar Bugs (list any bugs or tweaks for the sidecar below)

  • MAJOR: (Jul16/06) Any entries after button 18 dont save, they just disappear. Same goes when 2 extension units are connected. - joekane
    • NOTE: (Jul16/06) By using the Windows or Linux config file creator you can put these values in the fields. They will still disappear when you use the web interface after you download the cfg. - diver

  • MINOR (Aug12/06) Some lights on the extension unit do not consistently work properly. Several phone indicator lignts (3 out of 45) will go blank instead of Red when a call is picked up. These phone indicators will show blinking Red when being rung, so it is not the LED. One phone at position 15 does not show status at all. All phones were setup with the same configuration using a template. - diver

  • MAJOR: (Jul16/06) Red lights stay flashing when paging extensions setup as "Asterisk BLF". - joekane

Phone Bugs (list any bugs or tweaks for the phone itself below)

  • MAJOR: (Jul16/06) Display still blanks after some time. MAC 00.0B.82.03.XX.XX - julianjm
    • NOTE: (Jul18/06) It's funny because I've never seen my screen blank before but with this firmware a few of my phones did, however pressing the MUTE button fixed it as described in the previous firmware - ninthclowd
    • NOTE: (Jul19/06) One of my screens flipped upside down and had to be restarted. - diver
    • NOTE: (Jul20/06) On my phone blanking never went away. However, seems more common with this version. I've seen vertical scroll too, which then fixed itself immediately, and also blank when idle that refreshed on pressing '#'. Are we looking at a hw problem being worked around by sw or a recurring bug? - jedi98

  • MAJOR: (Jul16/06) Phone Book: When attempting to update phone book via TFTP or HTTP with XML file of format specified in 1.1.1.7 user's guide, phone freezes (clock stops, doesn't ring on call, totally unresponsive until power cycle) with "Sync Phonebook XML... This may take a minute" on screen. - acabtp
    • NOTE: (Jul16/06) I have the same issue, but it only seems to hang with more than 15 address book entries in the XML file. - gammacoder
    • NOTE: (Jul16/06) I am experiencing the problem with any sized XML file. If someone has an XML file that worked, please post it as an attachment so others may test. - acabtp
    • NOTE: (Jul16/06) Problem occurs with both PC and Unix formatted text files. Server logs indicate that the .xml file is successfully downloaded before the phone freezes. - acabtp
    • NOTE: (Jul16/06) The example in the manual worked for me. Copy and paste. Find it and a dynamic example at http://www.voip-info.org/wiki/view/GXP-2000+XML+Phonebook - Shane Steinbeck
    • NOTE: (Jul18/06) I couldn't get it do download without freezing untill I switched off the "Delete Manual Entries" now it's working a treat - Peter Almgill
    • NOTE: (Aug02/06) For me, anything between 15 and 20 entries is hit and miss. Sometimes it works, sometimes it freezes. Anything above 20 entries freezes every time. This is irrespective of whether "Delete Manual Entries" is enabled or disabled. - Channel-Two

  • MAJOR: (Jul16/06) The audio quality on all codecs is much worse with this firmware. The audio crackles (sounds like overmodulation), and adjusting both the codec and the TX frames per packet did not resolve the problem. Hopefully I'm missing an option that needs to be re-configured. - mansing23
    • NOTE: (Jul18/06) Agreed. I've noticed it as well - ninthclowd
    • NOTE: (Jul19/06) ditto - Anthony


  • MAJOR: (Jul20/06) Automatic Gain Control. Perhaps as an attempt to improve audio volume, Grandstream introduced an Automatic Gain Control (AGC) system. A distinct difference can be heard between older versions (more than a couple versions ago) and the newer ones, particularly that nearly every word you hear on the phone starts out quiet and gets loud towards the end. Short words like "no" spoken apart from any other words get nearly lost in the quiet. It is a poor implementation of an AGC, and even if it was not, any AGC that's noticable should be optional. This is not an acceptable approach to increasing the volume of the GXP-2000. I believe this qualifies as a major bug as this prevents this firmware version from being useful in a production environment.- thetatag
    • NOTE: (Jul20/06) First time I heard this effect I thought it was due to some heavy drinking the night before and ignored it lol. But you're absolutely right, this is definately annoying, and I don't believe it is worth keeping it, as the problems with the speakerphone remain. Speaking of the speaker phone... is it getting worse or is it just me? - ninthclowd

  • MAJOR: (Aug02/06) Vocoder Error. When "SRTP Mode:" is set to "Enabled but not forced" and you try to make an outgoing call you get an error on the screen saying "488 NOT ACCEPTABLE. Try a different vocoder." Seems to happen no matter which codec you choose. If it is not forced should it not fall back to RTP and allow the call to proceed. - naturalblue

  • MAJOR (Jul24/06) - Custom Idle Screen:Using custom bitmap with offset causes phone to crash. Suspect chunks of memory are being overwritten by the offset routine. The problem reoccurs every reboot when the phone attempts to load the Custom SCR. To remedy this, I had to reboot the phone with the network cable disconnected, and clear the custom screen from the preferences menu before it could be loaded. - acabtp
    • NOTE: (Jul31/06) I am using an offset of 3 for X and 18 for Y with no problems. Are you using a full screen image? - ninthclowd

  • MINOR (Jul24/06) - Custom Idle Screen:Random corruption on screen when using custom XML file. Noticable as dots or lines on the extreme left and right of text blocks. - acabtp
    • NOTE: (Jul25/06) This might be hardware or a problem with the way you encoded the image. I have 30+ phones that are displaying the same image with no corruption. Maybe its possibly a problem with the XML file corrupting during download? - ninthclowd
      • NOTE: (Jul28/06) There is no custom bitmap, only text fields in the configuration files I am using. I will post a picture on Monday. - acabtp

  • MINOR (Jul24/06) - Custom Idle Screen:Using the $d variable reference causes the phone to display only the last digit of the day of the month, and the rest of the <DisplayStr> after the $d is truncated. - acabtp

  • MINOR (Jul24/06) - Custom Idle Screen:The phone seems to always evaluate the "a1reg" as true, resulting in any <DisplayStr> that have "a1reg=false" never being displayed, and <DisplayStr> that do not have "a1reg=false" always being displayed, regardless of the Line 1 registration state. - acabtp

  • MINOR (Jul25/06) -Jitter buffer:I noticed the receive audio latency grows to around 100-200 ms (over approx 1-2 minutes) and never shrinks. There is no significant jitter on my system as everything is on the same LAN. I think something in the jitter buffer in the phone is not correct as I don't see this behaviour on other SIP devices on the same network.

  • MINOR: (Jul16/06) * AND ** do not work for speed dial. I use * for intercom (yes I know you can train the users to just press *, but they actually want a button labeled) and ** for Transfer to VM. If I program an extension with the * it works; but either by themselves just resets the state of the phone and I have seen the screen flash ** for the speed dial content of **. If dial manually it works. Any one know what the phone sends when you press a speed dial? - diver
    • NOTE: (Jul18/06) I might be wrong, but I'm pretty sure that if you put an extension in the speeddial buttons it gets prefaced with two asterisks. I wonder if when your putting the * into the speeddial it's getting translated to ***? If that was the case it would be possible to just add a *** and **** extension in the asterisk dialplan that rerouted to the original context. Just a thought O.o - ninthclowd
    • NOTE: (Aug11/06) I have worked on this some more and have determined the problem lies with the speed dial function. This did work for one week before upgrading to .13 or .16 (I don't remember now which one). The latest firmware sends the call (which is appropriate when you think about it). What I want to be able to do is have a field in the web/cfg that toggles whether or not the call is sent immediately after the speed dial button is pressed. I want to be able to say dial this number (in my case ** or *) then wait for user input before the call is sent. Yes, users should be able to press two buttons to send a call to voicemail, but it is very nice to have a button that is labeled "XFR/VM". - diver

  • MINOR: GUI-style menu system still has low usability. Colors are inverted for no obious reason. "Del"-key is delete and not backspace, which would be expected. No menu shortcuts on keypad. No keys which represent "ok" and "cancel" (have to navigate with arrow keys). - job
    • NOTE: (Feb07/06) Perhaps the up and down silkscreened/segmented arrows at the right of the display might be useful here? IMHO, simplify the GUI as much as possible, expand the text/list area to the whole screen width, and by dumping some UI you can get another line of text vertically. -Helix
    • NOTE: (Feb07/06) IMO it should be clean, uncluttered, no need to emulate windows, no need for scroll bar arrows. -Jedi98
    • NOTE: (Feb11/06) Agreed. There's no need for buttons and scroll bars. Something like this would be ideal for the menus � the user can select an item just by pressing a number (no need to use up/down, and options off the screen can be accessed w/o scrolling). The down arrow to the right of the LCD is on to indicate there is more beyond what is on the screen. Also, since the left arrow button goes back to the previous menu, the right arrow button should function like the select button. Similarly, when in a configuration screen, a simple layout like this for a phone book entry would be much easier to navigate. Up/Down Chooses a field. Press the Select or Right button to edit that field. When editing, the cursor is either an underscore, or maybe reverse text (better than the current �|� type cursor that makes the text next to it unreadable). Delete/Mute should act as backspace, just like it does when dialing a number. Pressing Select again finishes editing that field, and the user can use up/down/numbers to select the next field. For a toggle field, select toggles the value. -Ted
    • NOTE: (Feb11/06) Thank you. This is exactly what I meant with my first comment! You illustrate it perfectly. Also note that the Ok/Cancel concept is absent from your pictures which is a good thing. When a parameter is changed it is changed. No need to Ok changes. - job
    • NOTE: (Feb12/06) For the record- this is also exactly what I was talking about. The interface in your images is clean, efficient and uncluttered, and leaves much more usable display area. - Helix
    • NOTE: (Jul29/06) GS: Please let the users design the GUI for you!!. - jedi98

  • MINOR: (Jul3/06) Web UI Caching: Whenever I go the configuration web interface the data in the forms are always stale and I have to force a reload to get the current data. This is reproducable in firefox and IE using an ISA 2004 proxy cache. Recommend using the Cache-Control: no-store header instead of the pragma to fix this problem. -peter

  • MINOR: (Jul20/06) Download SCR XML fails after downloading it about 3 times in a row. This can be fixed with a reboot -ninthclowd

  • TWEAK: (Aug14/06) It's really nice to have SRTP now - however, with SDES the master password is sent unencrypted in the SIP stream AFAIK, so for real security it might be useless... Therefore the GXP-2000 should also support SIPS to make this solid (or is it already supported?) - Mirak

  • TWEAK (Jun05/06) built in microphone When you use the handset and press the speaker button the microphone should not switch to the build in microphone but should stay in the handset. This function is used for letting others listen to a conversation without notice, when the handset is on hook the built in microphone should be used. Right now its not possible to let someone listen to a conversation without lower quality for the called party, even old ISDN phones have this feature.- Datu

    • NOTE: (Jun05/06) Absolutly not! I've never seen a phone behave this way and it would drive me nuts. - nezer
    • NOTE: (Jun06/06) Agree w/ nezer... this is a very odd feature. Perhaps it's common in Europe? I live in USA and i have NEVER seen any phone act this way (and I've seen alot). I suggest a better way of listening is with asterisk and ChanSpy() to listen from another phone. Either way, if this is added it should be optional and disabled by default (If I found a phone that acted in this manner i would report it as a bug.). - Helix
    • NOTE: (Jun06/06) if you want hands free speaking you just let the handset on hook, this is one of the biggest complains with my users, when you want your office mate to listen to a support hotline or client while talking this is very useful. - datu
    • NOTE: (Jun06/06) I Agree. In Europe speakerphones work like this: If you press the "speaker" button and keep the handset off hook, you talk through the handset and you listen both through the handset and the speaker; if you put the handset on hook, you talk and listen only through the speakerphone. And I think this is a nice feature that should be implemented.. - Kurgan
    • NOTE: (Jun06/06) Agree w/ nezer as well... And I do believe this should be listed as a TWEAK and not a MINOR bug as this is not a bug at all since the phone is operating as intended. - ninthclowd
    • NOTE: (Jun16/06) Agree w/ ninthclowd. There's no bug here. It's a difference of oppinion on design. This should be a tweak and the logic should be configurable if possible. I have always found speakerphone logic to be very variable between phones, European or not. - jedi98
    • NOTE: (Jun22/06)I'm calling it. Changed this to a feature request since it seems to me the current behavior doesn't need to be tweaked, but the option to change it should be added as a new feature. Feel free to change if you think otherwise. -NateBell
    • NOTE: (Jul26/06)I cant really live with this bug, and its a bug, when you try to use the hands free mode its not useable, its very low sound, the other party cant almost hear you , its sound like far away, i could rather lay the handset to the desk hand have same results as this hands free mode. Anyway if i hold the handset in my hand, I want the mic in it to be used of course, please correct this bug Grandstream! If any of you think the internal mic should be used then just leave the handset on hook and use it. - Datu
    • NOTE: (Jul26/06) Just because something is listed as a bug or not does not necessarily mean it will or will not get fixed. Classification of a bug is only to let Grandstream know that their product is not working AS THEY INTENDED. The speakerphone mic and handset are working as they designed it (minus the gain issues which ARE bugs and the reason the handsfree mode is so ineffective). Thus what you are asking for is a tweak to change the product not a bug. I personally agree that this could be useful feature and many other people may have use for it, and it would also provide a bandaid for the speakerphone bugs. But if we label everything as bugs instead of tweaks, then grandstream will either A) start ignoring our classification, or B) implement new features without fixing their bugs with the product first. I don't want to start a flame war, so I won't change the classification. I personally think this WIKI should have two classifications: Type (i.e. Bug or Feature Request) and Severity (which could indicate how much the feature is wanted or how severe the bug is). Just stating my opinion =) - ninthclowd
    • NOTE: (Aug04/06) Again, this is a tweak, not a bug. Not even a minor one. - nezer

  • TWEAK: (Jul26/06) Instant Messages Should appear on the screen directly, line by line , so they can send be seen instantly without navigating through a menu. We want to send messages like 'calls forwarded to ...' or 'ringgroup' and see this immediately as instant messaging works like that- Datu


  • TWEAK: (Jul24/06) Getting Parked Calls to work right Works almost perfectly - read on. It could be downgraded to NO TWEAK or eliminated but the notes might be useful to someone. - Anthony
    • NOTE: (Jul20/06) I can confirm that pressing a "Parking Presence" button does not issue a **XXX command, whereas "Speed Dial" and "BLF" do. I have downgraded this note to TWEAK until I better understand the interaction of the Asterisk code, the metermaid patch and the GXP 2000 firmware. - Anthony
    • NOTE: (Jul19/06) The current firmware allows a BLF presence (steady lamp not blinking) to show a parked call but no more. My users want to see a flashing "Parked Call Presence" light/button, press it and retrieve the call. It then resembles a hold button for all extensions. This is what I think many wired current systems have. The current Asterisk method of throwing calls into a parking lot is not going down at all with my users. They refuse to accept this notion. While this is not just a GXP-2000 issue, it is mostly an Asterisk issue - I think the Grandstream firmware should let you pick up the handset, and press a "flashing light" button to pick up a parked call. That means the button/light should not just be a pretty face but be an active button. See my comments above for Asterisk 1.2.9.1 and trixbox 1.1 - Anthony
    • NOTE: (Jul19/06) When you press a flashing button, the phone makes a cal to ** (star star) followed by the extension you're monitoring. Just add dialplan logic to handle that. - julianjm
    • NOTE: (Jul19/06) Yes, I am aware of that. However, I will go back and check it. When you lift the handset and press the button, I dont see any interaction with Asterisk so I am assuming nothing is sent. Furthermore, its not a blinking light when a parking call is present and when the call is finished the light stays on. - Anthony
    • NOTE: (Jul20/06) Its not a blinking light because "show hints" on the CLI says the parked line 71 is "unavailable". Perhaps the metermaid patch did not make it into the trixbox 1.1 install of Asterisk 1.2.9.1 that I am using. - Anthony
    • NOTE: (Jul24/06) I switched to installing CENTOS 4.3 and Asterisk 1.2.9.1 with the metermaid patch from Asterisk 1.2.7.1 which compiles fine on 1.2.9.1. I used firmware 1.1.1.7 for GXP-2000. Set parkedhints=yes after installing the metermaid patch. Program the GXP for BLF on LINE 5 for extension 701, restart Asterisk with /usr/sbin/amportal start, and reboot GXP to register, and you get a steady light because Asterisk "show hints" says 701 is UNAVAILABLE. After the first parked call this light turns off since "show hints" correctly shows IDLE. Asterisk can then be set to understand the BLF button **701. Call parking works as (I) expected. - Anthony

  • TWEAK: (Jul20/06) Download SCR XML should download automatically on reboot if the option is configured, similar to the phonebook xml - ninthclowd


  • TWEAK: (APR02/06) Missed Calls Notification should go away after reviewing them, without having to clear the list. I would like to be able to clear the notification, and then revisit the missed calls list at later points in time to call the people back without having to write the phone numbers down. - Mike
    • NOTE: (JUL13/06) I Agree 100% Mike. Our system uses a ring all queue so missed calls pile on throughout the day. I personally wouldn't complain if there was a way to disable caller ID all together myself. - Mr Esteban
      • Comment: (JUL18/06) Mr Esteban, you can disable missed call notification under the Account information in the web GUI if you want to. I had the same problem, you get 25 calls per hour in a ring group and eventually you'll crash the phones that don't get answered. - ninthclowd

  • TWEAK: (May18/06) Blocked BLF Button. When a BLF is lit(remote user is on the phone) the phone silently ignores when the user presses the corresponding button. What If the user wants to leave a voicemail message for this person or maybe show up as "Call waiting"? Another example is that I use app_devstate from the bristuff package and I use it with my "global DND" function for all my phones. This block also denys me the ability to turn that DND function on and off with the same button. If there must be such a block it should be possible to disable it in the configuration. - Falle

  • TWEAK: (Apr04/06) Dialing from Call Lists. Once you have highlighted a number in one of the call history lists, it would be nice to be able to simply hit SEND to call it, rather than having to hit the little round button to go into another menu, press down until you highlight Dial, and then press the little round button again. On the same note, it would be nice to be able to highlight a number and press DEL to remove it. - thetatag

  • TWEAK: (May10/06) You still can't scroll to the end of a list by hitting the up button, for example on the menu, you have to scroll all the way to the bottom, it would be easier to just hit up to go to the bottom. - mike240se

  • TWEAK: (May14/06) "Do not Disturb" Mode. I'd like to have a better visual feedback for the Do-Not-Disturb-Mode. Now, the DND icon blinks in the display - but if the display backlight turns off, this is almost invisible, especially when not sittig directly in front of the phone. Ideally (in my opinion), the Message Idication LED should flash slowly (very slowly, maybe one short flash every two seconds, to distinguish it from the 'you have voicemail'-flashing!) if the phone is on DND mode. That way, I wont leave the room with the phone on DND, coming back later and forgetting to deactivate DND mode any more ;) - Phloogzoyk

  • TWEAK: (May23/06) Backlight Timeout. I for one would like to see a backlight timeout feature, so that the screen would stop illuminating after 3 minutes of inactivity if you missed a call or something. It would also be nice if the menu system had the same feature. So that if a user accidentally hit the menu button, the menu would exit after 3 minutes of inactivity. Just to preserve screen lifetime. - ninthclowd
    • NOTE: (Jul17/06) I use one in the bedroom and several others through the house. Having the backlight stay ON for a missed call is actually nice as I can see that we need to check the system. There is now a timer to extinguish the backlight after 4-5 seconds of hanging up a call (or clearing the missed calls list). Keep that one in the code - its really nice. Being able to set the timer in the configuration would be even nicer :) - mavila
    • NOTE: (Jul18/06) That may be the case in a low call volume environment, however I have 40+ users and some of them go on vacation or miss a call right after they leave, leaving the backlight for hours at a time. IMHO I believe there should be option to turn off the backlight via timer, just to protect larger investments - ninthclowd









Firmware Notes (Stable 1.1.0.16):


Currently available from Grandstream's Beta Site. This is a browsable directory, more stuff available at Grandstream Beta Site Home.

NOTE: 1.1.0.16 has been moved from the BETATEST directory to a full release version in DOWNLOAD/FIRMWARE ... the betatest directory now contains the preview version. numbered 1.1.1.7, not 1.2.*.* as was suggested earlier.
NOTE: The Users Manual at http://www.grandstream.com/user_manuals/GXP2000.pdf has been updated for 1.1.0.16 as well. -kam

Changelog
Build 1.1.0.16

Fixed IP Fragmentation bug
Fixed GXP-2000/BT500 ring tone change via keypad menu not effective after reboot

Improved audio quality with some audio parameter changes
Fixed we crash on attended transfer on platforms that use To/From headers without square brackets
Fixed we reject cfg files smaller than 512 bytes
Fixed BT-200 keypad UI for TFTP server not working
Fixed a typo in LCP (PPPoE related)
Fixed GXP2000 provides RING only for first incoming call; when first caller hangs up, the ringing stops.
Fixed BT-200/500 does not play short beep on auto-answer
Fixed Bug in Via header when DNS name is used instead of IP address
Fixed we still responds "recvonly" on un-hold SDP message

Changelog listed above taken from Grandstream 1.1.0.16 Release Notes PDF -Helix


Bugs / Tweaks (1.1.0.16):


  • MAJOR(Aug02/06) Bug in IP implementation: The firmware incorrectly treats certain IP address as a local broadcast address which prevents it from communicating with devices with such IP addresses. It looks like the firmware does this: if dest_ip & ~netmask) == ~netmask) it_IS_broadcast; instead of this: if (dest_ip == (my_ip it_IS_broadcast; For example, if your IP address is 1.2.3.162, netmask 255.255.255.224 and gateway 1.2.3.190, a packet to 4.5.6.31 should go through the gateway, but the phone considers it a local broadcast and sends it to ff:ff:ff:ff:ff:ff ethernet address. The longer your netmask is (i.e. the smaller your subnet is), the more destinations you will be unable to reach. I have reported this bug to support@grandstream.com when 1.1.0.13 was the latest stable and again when 1.1.0.16 came out, but I have not received any reply (except for the "ticket generation") from them. Jaroslav Janacek

FIXED 6/29/06 The low volume on the speakerphone when paging is fixed. The new volume level works fine for me. Anthony ___

  • MAJOR: (Jun19/06) Mute after putting a line on hold: If you are on a call and you put the party on hold, then you recieve a second call on another line but don't answer, when you go to take line 1 off hold they will not be able to hear you. Please note that I built a rollover macro(in asterisk) so you can use one SIP account for all lines on the Grandstream and new calls will come into any available line even while your on the phone. It's possible this could be a software setting but I believe this to be hardware, especially because of the workaround listed below. - ninthclowd
    • WORKAROUND: Tap the handset release button and they will be able to hear you again. - ninthclowd
    • NOTE: Why do you need a macro to receive multiple calls on a single account? GXP-2000 can handle up to 11 calls on a single account without any special setup. Or maybe I have not understood what you are doing. - Kurgan
    • NOTE: (Jul03/06) It's so Asterisk will still dial the phone even though it's reporting BUSY. I believe this is not really relavent in retrospec - ninthclowd

  • MAJOR: (Jul01/06) Very soft outbound volume through handset: Apart from the problems with the speakerphone, the volume of the caller to the callee is extremely soft when talking via the handset. I test by calling myself via GXP2000 -> Asterisk (SIP - g729 or alaw) -> VSP (alaw) -> PSTN and placing my PSTN phone on mute while speaking into GXP2000. I have tried various providers and the voice level is the same low output using GXP2000. When I make the same test calls via SPA3K the volume is perfectly loud and clear. I don't even find the speaker volume on GXP2K increases the mic level. I found this when doing testing with firmware 1.1.0.13 and find it continues with this version. I hope individual RXGAIN/TXGAIN is on the way! - Giulio
    • NOTE: (Jul03/06) I agree. This is a must fix. - Anthony

  • TWEAK: (May10/06) You still can't dial a number when the phone is on hook by dialing the number then hitting send or speakerphone. - mike240se
    • NOTE: (Jul03/06) On hook dialing has been open for some time in the feature request section, please post any comments on this topic there as to keep everyone on the same page. - ninthclowd

  • MINOR: (Jul5/06) UI: When picking up line appearances 10 or 11 the display shows LINE:: and LINE;: instead of LINE10: and LINE11:. - Andrew

  • MINOR: (Jul18/06) BLF: When you have an external number programmed into speed dial key 1 configured as a normal button and then have subsequent buttons configured for BLF the phone tried to register the number associated with key 1 as BLF also. First noticed with 1.1.0.13. Currently have a ticket open with grandstream. - Gareth

  • TWEAK: (Jun30/06) Add support for SIP header Alert-Info to trigger custom ringtones, e.g. SIPAddHeader("Alert-Info: ring3"). This was a feature request and can be seen further below on this web page. I decided to stick my neck out and bump it up since the absence of this feature bugs me. Feel free to shoot me down. I would like to see an easy way to distinguish internal vs external calls via ringtones on the same extension line. It seems to me the best answer is SIPAddHeader - are there other easy solutions? - Anthony
    • NOTE: I strongly agree! This is a feature I am been waiting for since when I bought my first GXP-2000. Many of my customers want this! - Kurgan
    • NOTE: This may already be possible (Jul 01/06) According to a Grandstream FAQ, this is already possible. Set the 'distinctive ring number' for rings 1-3 to a word (soemthing thats not a number), then send that word as alert-info. I will try this if i get a chance later, but it would be very cool if it were true! -Helix
    • NOTE: It works! Use something like this: exten => 12345,2,SetVar(ALERT_INFO='<ignored text>\;info=ring_word'). Then you have to put the "ring_word" as the distinctive ring number in the phone's config. It would be even better if the phone could simply recognize "ring1", "ring2" and "ring3" as Alert-Info values, but it does not. You have to set up distinctive ring strings in phone's config to make it work. - Kurgan
    • NOTE: It works! The above did not work for me, but the following did: exten => 12345,2,SetVar(_ALERT_INFO='<ignored text>\;info=ring_word'). I use A@H 2.5 which has Asterisk 1.2.0 .. - Anthony
    • NOTE: (Jul03/06) This is not a bug. Relabeled as a tweak. - ninthclowd
    • NOTE: (Jul04/06) Setting variables depends on Asterisk version, I forgot to mention that I use Asterisk 1.0.7. - Kurgan
    • BUG? (Jul04/06) It seems that only ringtone 2 works using alert_info. If I try to use ringtone 3 I get the default ringtone (1). So I end up with TWO ring tones (default and ringtone 2) and not three. Could someone check and report back? - Kurgan
    • BUG? (Jul05/06) I agree. For me works only custom ringtone 1. So only 2 ring tones are available (even better than 1 :-) ) - maxx

  • TWEAK: (Feb24/06) Speakerphone volume setting should be separate from handset volume. I need to turn the speakerphone up to full volume to hear it, but this means the handset is too loud. - bani

    • NOTE: (Sept04/06)
If you have two numbers registered on the phone can one be made to glow green and one made to glow red depending which extension is ringing. The reason for asking, If you have two company names running on one phone, then when more than two calls dome in you cannot distinguish which line is for which extension as the last call always shows on the display and all lights are red.









Firmware Notes (Beta 1.1.0.13):


Currently available from Grandstream's Beta Site
  • NOTE: (May18/06) The above link seems to be broken. Anybody know where to get it? - ninthclowd
  • NOTE: (May18/06) Link fixed. Grandstream changed one of the folders on their site. The new folder (/BETATEST/GXP2000_BT200) seems to suggest perhaps an upcoming product? Another file in the folder has firmware for bt200 and bt500 products also... can't wait to see them! - Helix
  • NOTE: (May19/06) I didn't happen to see the release notes bundled in this package. Anyone have a copy they can post? - ninthclowd
  • NOTE: (May19/06) The site is a browsable directory, click here and you can find all sorts of cool stuff. The files you want are in the GXP2000_BT200 folder. -Helix
    • NOTE: (May22/06) Thanks for the link! -ninthclowd

Changelog
Build 1.1.0.13 5/16/2005
  • Add Quick IP Calling mode
  • Fixed GXP-2000 Speed Dial/Asterisk BLF pick up broken in 1.1.0.12

Build 1.1.0.12 5/15/2006
  • Fixed GXP-2000 crashes when a very long DTMF string is dialed
  • Fixed SIP NOTIFY to event REFER violating RFC 3515
  • Fixed we do not affix To-tag for PRACK request
  • Fixed we do not use new branch for PRACK request
  • Fixed we do not include Contact header in 180
  • Fixed we do use random port for RTP even if random port is set to yes

Changelog listed above taken from Grandstream 1.1.0.13 Release Notes PDF included with 1.1.0.13 firmware -benny23


Bugs / Tweaks (1.1.0.13):

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!
(In addition to adding user feedback here, please copy items from the previous bug/tweak list to this list if you discover that they are still issues and note that you have done so! Thank you!)

  • MAJOR: (Jun19/06) Mute after putting a line on hold: If you are on a call and you put the party on hold, then you recieve a second call on another line but don't answer, when you go to take line 1 off hold they will not be able to hear you. Please note that I built a rollover macro(in asterisk) so you can use one SIP account for all lines on the Grandstream and new calls will come into any available line even while your on the phone. It's possible this could be a software setting but I believe this to be hardware, especially because of the workaround listed below. - ninthclowd
    • Workaround: Tap the handset release button and they will be able to hear you again. - ninthclowd

  • MAJOR: (Jun13/06) Speaker phone does not work at all and hangs when doing a soft reboot (from the web server). MAC is 00.0B.82.09.75.13 - Drew
  • MAJOR: (Jun09/06) When using the GSM codec, silence supression is always on, regardless of the setting in the options. - Adrian2k
  • MAJOR: (May22/06) FWIW, Screen still blanks! MAC is 00:0B:82:03:A1:25 - MattB
    • NOTE: (May22/06) In the 40 phones that my company purchased, one of them seems to blank. I'm wondering if maybe this is the result of a partially defective unit rather than firmware issues. - ninthclowd
    • NOTE: (May22/06) Once again my screen blanked. So far it has only happened once after 3 days. Look like the blanking is still an issue, but not as bad as it used to be. -NateBell
    • NOTE: (May25/06) This might help diagnose - my screen was blank again this morning. A call came in for me (whilst the screen was blank) and I hit MUTE/DEL to send to Voicemail. The second I pressed it the screen came back again! - MattB
    • NOTE: (May25/06) Screen blank workaround Further to my previous entry, whenever your screen blanks just press the MUTE button - brings the screen back every time. - MattB
    • NOTE: (Jun23/06) One of my phones with a MAC of 00:0B:82:03 blanks immediately after the Grandstream logo flashes by on bootup and nothing seems to bring the screen back after this happens. The screen works fine on the older 1.1.0.1 firmware (perhaps the 1.1.0.11 as well, but I haven't verified that) - Flu
    • NOTE: (Jun29/06) Phone with a MAC of 00.0B.82.03.D4.32 blanks, generally triggered by an incoming call. But not every call, this is intermittant. MUTE/DEL button does not seem to bring the screen back, only a reset will do it. The last version that worked (mostly) was 1.0.2.13 - Jedi98

  • MAJOR: (May22/06) Quiet Speakerphone. I might get flamed for this, but I am elevating this to a major bug. The speakerphone continues to be esentially useless. Yes, the echo problem it had under 1.0.1.9 has gone away, but it is too quiet to be usable in all but the most silent environment. If you have any ambient noise, the speakerphone experience is ruined. And even in silence, it is very quiet - even at the loudest settings. The speaker is certainly capable of producing much louder sound, as is evidenced by the maximum ring volume, and a good echo cancelling algorithm should be able to handle the added volume being picked up by the microphone. In my business deployment, this is the number one complaint about these phones. - thetatag
    • NOTE:(Jun14/06) I agree with this. I could get by with paging a phone for which I could set the volume. Unfortunately, a soft or hard phone reboot will reset the volume to default and render the page inaudible. I would like to see a programmable volume control which might be a first step towards a fix. - Anthony
    • NOTE: (Mar23/06) I'm going to have to agree. The single biggest complaint I recieve about these phones from my end users is about the speakerphone issues. It is in fact quiet, however i think the bigger problem is the lack of clarity. It seems like the speakerphone quality just gets worse and worse with each release (but at least there isn't any echo anymore) - ninthclowd
    • NOTE: (Jun21/06) I hate to backtrack but now after playing with it more, I believe the quality issues are directly related to the speakerphone being so quiet. Therefore I would like to state that the bigger problem is the volume. - ninthclowd

  • MINOR: (May22/06) Call Distortion: Since I updated with this firmware all phone calls sound distorted on my end, but the other party hears me fine. Reverting back to the previous firmware did not solve the issue, which leads me to believe that it may not be possible to revert back once this firmware is loaded. - Mike B.
    • NOTE: (May29/06)I also have this problem with the current firmware. -Richard H.

  • MINOR: State Bugs. Loopback call (dial phone from itself, via asterisk) drops out on answer (as expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up arrow adjusts the volume instead of missed call list. -Jedi98
    • NOTE: (Mar21/06) Confirmed the problem on the latest beta firmware "1.0.2.13". -ChrisUK
    • NOTE: (May22/06) Reproduced on 1.1.0.13. - job

  • MINOR: (Mar29/06) Numbers off-screen. When entering long phone numbers (i.e. for international calls) the phone number goes off-screen as it advances, rather than moving up to the upper line as it used to. - Mike

  • MINOR: (Mar29/06) Ring Volume. Ring volume still does not survive reboot - it resets back to middle setting. - Mike

  • MINOR: (Mar31/06) Muted DTMF. You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag
    • NOTE: (May24/06) Reproduced with 1.1.0.13. - vgster



Firmware Notes (Beta 1.1.0.11):


The firmware 1.1.0.11 is attached to this page:
Image


Changelog
Build 1.1.0.11 4/28/2006
  • Fixed the ping problem when the device is in router mode
  • Enabled the broadcast drop mode (this should improve the switch performance for multicast and broadcast)
  • Fixed the 3-way conferencing issue when the re-invite to bring the 1st party out of hold status gets challenged.
    • This is difficult to verify. Basically our 3-way conferencing can be broken into the following steps:
      • A invites B
      • A re-invites B to put B on hold
      • A invites C
      • A re-invites B to put B out of hold status
    • If at step d) the proxy challenges the request with 401/407, we couldn’t complete the conferencing
  • Fixed the PPPoE TCP problem
  • Added idle timers to fix more idle screen blackout cases
  • Fixes for blank LCD or corrupted GUI issues
  • A-Tick and DC filter changes
  • Fixed crash on incoming call when all channels are in use
  • Fixed lost registration problem
  • Fixed we will never switch DNS server even if primary DNS server failed to respond and there is a secondary DNS server
  • Changed for GXP-2000: once entering direct IP calling mode, the cursor focus is in the text field instead of CANCEL button
  • Reduce the GXP2000 handset earpiece audio level by 4.5 dB
  • Fixed GXP-2000 crashes when using MISSED CALL GUI to dial out
  • Added use of MUTE/DEL key during incoming call ringing state will reject call using 486
  • Added MUTE/DEL key will act as toggle key to turn DND on and off during idle
  • Fixed GXP-2000 direct IP call cannot specify port.
    • Note that a new input method is specified here: you will use * to enter dots (separator between octets) and use # to enter colon (separator for port). So you can enter "10.10.12.135:5068" using "10*10*12*135#5068". This is probably more intuitive
    • Note 2: Direct-IP calling feature is further cleaned out so that STUN mapped info is not used when we detect direct IP calling destination is in local subnet (From and Contact headers are also cleaned to use IP address only, not including the configured SIP URI).
  • Fixed AGC setup change
  • Fixed RTPSend bug
  • Fixed we do not use the previous SSRC, timestamp, and sequence number after restoring a previously hold call
  • Fixed static IP problem in 1.1.0.2
  • Fixed we start sending RTP when restoring a call before we receive 200 OK Fixed we do not clear out CallFwd settings when user configure to disable call features
  • Added special factory workaround mode-when configured to use static IP 192.168.0.160 and no gateway IP is configured, provisioning is skipped
  • Added support for Broadsoft Click-to-Answer feature using "talk" event
  • Fixed GXP-2000 cannot make direct IP calls
  • Fixed GXP-2000 factory MAC-Edit function cannot change last digit to A-F
  • Fixed redial does not append the dial-plan prefix
  • Fixed we will retry 5 times if only config server is configured and there is config file
  • More LCD fix for GXP-2000
  • Fixed some crashes Issues

Changelog listed above taken from Grandstream 1.1.0.11 Release Notes PDF included with 1.1.0.11 firmware - flu



Bugs / Tweaks (1.1.0.11):

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!
(In addition to adding user feedback here, please copy items from the previous bug/tweak list to this list if you discover that they are still issues! Thank you!)


  • MAJOR: (APR17/06) Screen still blanks! MAC is 00:0B:82:03:A1:25 - MattB
    • NOTE:(May05/06) Updated: I can confirm my phone's screen blanked this morning. My MAC address is 00:0B:82:03:A0:??. -Nate Bell
    • NOTE:(May06/06) My phone does *not* have blanking problems. Mac 00:0B:82:05:AA:xx - Kurgan
    • NOTE:(May08/06) blanlking is now mostly corrected .. went blank during a call a few times, but mostly is now OK. Previously was just blank all the time or upside down etc, so a BIG improvement. - Robin Sz
    • NOTE:(May08/06) Just to clairfy, with 1.0.2.13 my screen would blank after a couple seconds upon powering on. I reverted back to 1.0.2.8 where my screen would blank after a day or so. I still have blanking issues with 1.1.0.11, but so far it has only happened once, and I've been running the firmware since it was released. So I'm still having a problem, but as Robin said, it has been much improved for me. -NateBell
    • NOTE:(May12/06) (possible hardware issue) Since 1.0.1.11 my old gxp (mac 00.0B.82.03.CD.xx) LOCKS up... a LOT. Used to blank once every few days, now it locks up (screen is frozen but image still present, clock stopped) several times per day, often within a few minutes of powering it on. Tried factory defaults, flash back to 1.1.0.1 w/ no luck. I think my phone may just be dying, but figured I'd toss this in anyway in case it helps. -Helix
    • NOTE:(May12/06) I can confirm my phone's screen blanked too. My MAC address is 00:0B:82:03:CC:2E. -cervajs
    • NOTE:(May17/06) Blank screen too. My MAC: 000B8203CC2F. -Festr

  • MAJOR: (May 3/06) Still problems with BLF in Firmware 1.1.0.11. No lights blink or lighten up when other partner is calling or getting a call. This was fixed in a very previous version 1.0.2.3. I tested with two GXP2000. I can see in Asterisk CLI (show hints) that the hints are active and working but the Watchers still 0.- FreshSmith
    • NOTE: (May04/06) Do you have an old GXP? My phone blinks and lights up like an x-mastree when other users are on the phone. - Falle
    • NOTE: (May04/06) Seems ok on mine - ninthclowd
    • NOTE: (May05/06) Mine is ok too. What are the MACs on your units? - qortra
    • NOTE: (May05/06) This isn't much help, but mine are working too. -Nate Bell
    • NOTE: (May06/06) Mine works too. Mac 00:0B:82:05:AA:xx, Astersik 1.0.7 (Debian Sarge) - Kurgan
    • NOTE: (May06/06) Sounds more like a config error, did you set subscribeextension= in sip.conf? - mike240se
  • MINOR: (May05/06) Ring Volume still not survive to a reboot, but now, the default volume is lower... - Snipefoo
  • MINOR: (May05/06) Ring Tone needs a reboot when setting from the phone menu, but not when setting from de web UI - Snipefoo
  • MINOR: (Mar31/06) Muted DTMF. You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag
  • TWEAK: (May04/06) Blocked BLF Button. When a BLF is lit(remote user is on the phone) the phone silently ignores when the user presses the corresponding button. What If the user wants to leave a voicemail message for this person or maybe show up as "Call waiting"? Another example is that I use app_devstate from the bristuff package and I use it with my "global DND" function for all my phones. This block also denys me the ability to turn that DND function on and off with the same button. If there must be such a block it should be possible to disable it in the configuration. - Falle
  • MAJOR: (May05/06) Call mute when switching lines. When using multiple lines for one extension (i.e. for rollover purposes) the phone will sometime mute one or both of the parties when switching lines. At least I assume it's a mute. This was not an issue in 1.0.2.13 and previous - ninthclowd
    • NOTE:(May06/06) Could you please explain in a more precise way what do you do to produce the bug? - Kurgan
    • NOTE:(May09/06) I'm not exactly sure what's causing it, but I&#039;ll tell you how I'm setup. I set up line 1 to have a SIP account. Line 2,3 and 4 I leave default. Then when I'm on a call on line one, I can dial out on line 2 but the remote party will sometimes get muted. They will be able to hear me but I can't hear them. The call continues and the timer still counts and I can switch to and from line 1. I find that it seems to happen when using the line buttons to switch between lines. - ninthclowd
    • NOTE:(May09/06) So we're talking loss of one/other/both RTP stream maybe. One idea, and it's a real stab in the dark, is the codec being re-negotiated? If any of it goes via Asterisk you can probably get verbose messages from there. What about the syslog? - Jedi98
    • NOTE:(May10/06) What should I be looking for in the log? I did find some of these "NOTICE[6998] rtp.c: Unknown RTP codec 100 received" don't know if thats normal or not but I don't remember seeing them before. - ninthclowd
  • FEATURE: (May14/06) I'd like to have a better visual feedback for the Do-Not-Disturb-Mode. Now, the DND icon blinks in the display - but if the display backlight turns off, this is almost invisible, especially when not sittig directly in front of the phone. Ideally (in my opinion), the Message Idication LED should flash slowly (very slowly, maybe one short flash every two seconds, to distinguish it from the 'you have voicemail'-flashing!) if the phone is on DND mode. That way, I wont leave the room with the phone on DND, coming back later and forgetting to deactivate DND mode any more ;) - Phloogzoyk





Firmware Notes (Alpha 1.1.0.4):


This is an internal build correcting issues in the current Beta. Distribution of this build has been prohibited by Grandstream. (Expect to see a public release version soon.) - thetatag

Changelog
Build 1.1.0.4 3/24/2006
  • Fixed GXP-2000 crashes when using MISSED CALL GUI to dial out
  • Added use of MUTE/DEL key during incoming call ringing state will reject call using 486
  • Added MUTE/DEL key will act as toggle key to turn DND on and off during idle (These two are the sweetest part most people like)
  • Fixed GXP-2000 direct IP call cannot specify port.
    • Note that a new input method is specified here: you will use * to enter dots (separator between octets) and use # to enter colon (separator for port). So you can enter "10.10.12.135:5068" using "10*10*12*135#5068". This is probably more intuitive.
    • Note 2: Direct-IP calling feature is further cleaned out so that STUN mapped info is not used when we detect direct IP calling destination is in local subnet (From and Contact headers are also cleaned to use IP address only, not including the configured SIP URI).
  • Fixed AGC setup
  • Fixed RTPSend bug
  • Fixed we do not use the previous SSRC, timestamp, and sequence number after restoring a previously hold call

Bugs / Tweaks (1.1.0.4):

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!

  • MINOR: (Mar31/06) Muted DTMF. You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag

  • TWEAK: (Mar31/06) MUTE/DEL Call Refusal. Currently, if you press MUTE/DEL while the phone is ringing, it immediately responds as busy. However, if the phone is ringing while you are on the phone (call waiting scenario), the MUTE/DEL button mutes your current call. Personally, I would like to see it also respond busy even if you are on the phone when you press it. How likely is it that you will want to mute you current conversation while another call is ringing in? It seems more likely that you would want to refuse a call-waiting call without having to hold your current call. This is subjective, though, so I'd like to see a little feedback here before recommending this behavior to Grandstream. - thetatag
    • NOTE: (Mar30/06) How about: press MUTE/DEL quickly to mute the call, hold it down (more than 1 second) to toggle DND status while on a call. - bani
      • NOTE: (Mar31/06) The MUTE/DEL button now has three functions. It toggles DND status (when the phone is idle), it refuses an incoming call (when the phone is on-hook and ringing), and it mutes the current call (when the phone is off-hook and in an answered state). I like your idea, except I would prefer the longer button-press toggle DND when no call-waiting lines are ringing and reject an incoming call if there is a call-waiting line ringing. Although, personally, I only see marginal value in being able to toggle DND status while on the phone. - thetatag

  • TWEAK: (Apr03/06) Daylight Savings Time. Daylight savings time is implemented in the GXP-2000 by simply setting the time an hour later when the option is selected. This results in having to change this setting on every phone (or a global configuration file) and rebooting every phone to update the time. (The time may update without a reboot, but it does not do so immediately.) The GXP-2000 should have daylight savings time implemented like nearly every other smart system: it knows when to turn it on and off based on the date. The option, then, should be "Automatically adjust for daylight savings time," and the phone should take care of changing at the appropriate times automatically. - thetatag
    • AGREED! - Kurgan
    • Ya manually changing DST on 40 something phones was not that fun. - ninthclowd
    • NOTE:(Feb04/06) I am a bit skeptical. If this cannot be done reliably it should not be done at all. Can Grandstream input the local rules for the whole world? How can the phone know Australia postponed DST this year? But if they let you edit the DST rules manually it is a good idea. My computer does this reliably so perhaps it is possible. - job
    • NOTE: (Apr04/06) NTP handles daylight savings time. - bani
    • NOTE: (Apr05/06) This won't apply to all of us, but according to a thread on the AAH forums, you can have the gxp-2000 get time from an AAH server by placing the servers IP into the gxp's NTP setting (NTP is setup and enabled by default on AAH, and likely will work on a straight Asterisk installation if NTP is enabled as well). - jehowe
    • NOTE: (Apr06/06) What do you mean NTP "handles" DST? NTP (or actually SNTP here) synchronizes time. That time is UTC time. How the client interprets that into the displayed time which must be corrected for time zone and DST has nothing to do with the synchronization protocol. - job
    • NOTE: (Apr04/06) Err yes you are correct. Hardcoding rules for every locale on the planet is not practical, so grandstream should just put in options to configure the start and end dates for DST. Even DST for the US will be different in 2007, thanks to the Energy Policy Act of 2005. - bani
    • NOTE: (Apr27/06) You can do this centrally on DHCP (presuming you have admin of your DHCP server) by setting the time offset (option 2) and allowing it to override the TZ on the phone. Also set the phone DST off. Then you only need to change the DHCP time offset on the time change. - jedi98
    • NOTE: (May03/06) Good idea, it has also been implemented in beta 1.1.0.11, but won't this break other clients (not phones) that get timezone from DHCP? I mean, here I am changing timezone, not DST settings. Won't other clients get a wrong timezone and get confused? - Kurgan
    • NOTE: (May08/06) Good question. I tested this, admittedly only with GMT, but it has not caused a problem here. I think this is because time-offset, on DHCP, when used correctly by a client replaces the TZ rather than adding to it. You should be alright as long as all your clients are in the same TZ. However, as always, you should test before applying this to any critical environment. - Jedi98



  • TWEAK: (Apr04/06) Auto Call Waiting. In asterisk there seems to be no way of setting CW by default. Grandstream should add an option to have the phone run a set of commands just after registering...... such as *70 to enable CW (Call Waiting) or *71 to disable, *72 for CF (Call forwarding) or *73 to disable CF - naturalblue
    • NOTE: (Mar31/06) Why would you want Asterisk to handle Call Waiting? This is a multiline phone. I propose to remove this request unless better explained. - job
    • NOTE: (Apr04/06) This tweak request doesn't make any sense (and it's unsigned) so i'm moving it to the bottom. - bani
    • NOTE: (Apr04/06) I think this is a good idea. My GXP2000 has CW enabled and yet it still doesn't accept the next call in unless i enable call waiting in asterisk. As this phone is not just for simple 1 line connections and has been marketed as an enterprise phone i think the ability for it to easily integrate in PBX solutions such as asterisk is very valuable. Also it would allow for the phone to run multiple features in the future by simply running a command after registration. i have signed this post and my original, i thought the wiki would add it for you as a registered user. - naturalblue
    • NOTE: (Apr04/06) You must be using asterisk@home or something — call waiting is not something asterisk "does", and a phone's job should not be to enable/disable features on the PBX. The PBX should do that itself. This is not a feature that belongs in the phone, someone should fix asterisk@home or whatever you are using. - bani
    • NOTE: (Apr05/06) Same as bani said pretty much- for a SIP channel, * will always have call waiting unless it's turned off (possibly with a call limit). When a call is in progress on a SIP channel and another call comes in, * tells the SIP endpoint (ip phone) that another call is coming in (this is considered default behavior for SIP). The GS deals with this by playing a call waiting tone, printing info on the display, and making one of the other LINE lights flash (even if the call is not coming in on that button's account, the line it comes in on is displayed on the LCD). You can then if you want put your current call on hold and switch back and forth between the two calls. If * or your voip provider are not sending the second call, then they are misconfigured. Perhaps you have it set up like a fax line which usually shouldn't get CW? Or if you're using Asterisk@home perhaps it has a funky feature set... either way while dialing something on registration might be useful for some hacks, it's not required to make call waiting work. Hope that helps :) - Helix

  • TWEAK: (Apr04/06) Dialing from Call Lists. Once you have highlighted a number in one of the call history lists, it would be nice to be able to simply hit SEND to call it, rather than having to hit the little round button to go into another menu, press down until you highlight Dial, and then press the little round button again. On the same note, it would be nice to be able to highlight a number and press DEL to remove it. - thetatag

  • TWEAK: (Apr10/06) Some useful things It would be nice if phones would support that:
    • Call forward if no answer after settable timeout
    • Dialing from call list by just picking up handset or pressing speaker or send button(and maybe #)
    • Option to set busy trigger to limit max incoming calls at same time
    • set callforward call by pressing transfer button (instead of dialing *72) and number (onhook - transfer all, off hook - transfer if busy)
    • If down key is pressed, phonebook appears on the beginning. Why wouldn't up key display phone book on the end? Show missed calls could still stay up, because first you check missed calls, right? - bad2Dbone

  • TWEAK: (Apr16/06) DHCP Option 66. This option for allowing dhcp to override tftp should be set to on/yes by default. You can set the provisioning of this (using p145) in the cfg file for the phone but this is no use as the phone wont go to your tftp server to get the cfg file until you manually log into the phone and set this option to yes. If provisioning/setting up lots of phones this causes quite a delay. - naturalblue


(In addition to adding user feedback here, please move items from the 1.0.2.13 bug/tweak list to this list if you discover that they are still issues! Thank you! - thetatag)



Firmware Notes (Alpha 1.1.0.1):


WARNING:
(Mar29/06) Grandstream has stated that this firmware version was not authorized for public release. It was released at a work-in-progress state to a few individuals with particular firmware issues to verify that a change in one piece of code would correct their problem. If you choose to use this version, be warned that there may be known broken functionality, bugs, or unstable developmental code. You have been warned! - thetatag

This firmware rev was posted to the bottom of this page a few days ago and apparently nobody noticed. Unknown if distribution is sanctioned by Grandstream, assuming it's ok because it's been posted here for a few days. - Helix

Also note that you can revert from 1.1.0.1 back to 1.0.2.13 (software) and 1.0.2.3 (bootloader) should this cause problems for you. -Mike

The firmware 1.1.0.1 is attached to this page:
Image


Changelog
This build was supplied without a changelog. Perhaps someone can add it?
Attempts to fix early-model GXP display corruption issue are not succesful.
  • Added: two new options in the Special Feature drop-down of the account page: RNK and Sylantro. -Mike
  • Changed: default display screen now shows name and number instead of date (when time is in top right). (Please correct me if this was already included in 1.0.2.13, however, I never noticed it there.) - Mike


The ZIP also has two additional files- bt200a.bin and bt500a.bin. I don't think my GXP asked for them. Maybe these are related to the new sidecar, or upcoming lower-end phone models?
  • Note: The naming convention suggests that the other two files mentioned above are firmware releases for (upcoming?) BudgetTone phones 200 series and 500 series. -Mike

Bugs / Tweaks (1.1.0.1):

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!

  • MAJOR:(Mar30/06) Conference Call-MUTE. When in conferenced call unable to Mute phone. - MikeB
    • NOTE: (Apr03/06) Some months ago I posted the bug that muting while on a conference call caused one or both of the remote parties to not be able to hear the other (as opposed to just muting the microphone on the local phone). I wonder if rather than fixing the problem, they just decided to eliminate mute functionality on conference calls. If that's the case, it is certainly the wrong answer. - thetatag

  • MAJOR: (Apr11/06) Blanking Screen. Phone's LCD becomes blank after a few hours or so on a 00.0B.82.03.xx.xx MAC addresses phone, it seems to happen alot less (used to happen 3-4 times a day now its more like once every 1 or 2 days) than with 1.0.2.3 firmware but it is still happening, my phone rings for every phone call coming into our company (i dont answer it just rings and reception pick it up,) the other phone we are using doesnt do this and i havent yet seen that phone blank with this firmware... in other words the blanking problem is directly related to the number of calls sent to the phone (this was also the case with 1.0.2.13) . - SoloFlyer
    • NOTE: (Apr10/06) Just wanted to mention that I have not been experiencing this with mine on a 00.0B.82.04.xx.xx MAC phone. - Mike
    • NOTE: (Apr16/06) Worse than ever on my 00.0B series mac phone ... Display lasts about 3 seconds after reboot before going blank, regardless of whether plugged intot he network or not ... a backwards step again I'm afraid. How I wish I had never upgraded :( . - robinsz
    • NOTE: (Apr18/06) I just came back from a 4 day weekend and the phone hasnt blanked, but since there wouldnt have been receiving phone calls, i belive it confirms my theory that its related to phone use. - SoloFlyer
    • NOTE: (Apr21/06) My experiences with this firmware support the going theory. I have 2 phones running 1.1.0.1, one that gets a high volume of calls (10-20 per day) and one that gets almost no calls. The called phone usually has a blank screen in the morning, while the other phone doesn't. -Nate Bell

  • MINOR: (Mar29/06) 503 error. If the phone receives SIP/2.0 503 Service Unavailable it will print 503 in the display. It should print Unavailable instead. How to reproduce: make exten => 5555,1,Congestion in asterisk extensions.conf, then dial 5555 from the gxp2000. - bani

  • MINOR: (Mar29/06) NTP Broke NTP doesnt work at all and phone seems stuck on January 2nd at 12am everytime i look at it. Since there is no way to change the date/time manually, you just have to deal with no date and time. - mike240se
    • This may be an issue with your phone or the individual NTP you are trying to connect to. My NTP is working fine (try time.nist.gov). - Mike
    • I tried setting it to time.nist.gov and also to an IP, I also tried assigning it via DHCP (and changed the option to allow it in the web gui) and no go, stuck on january 2nd. With 1.0.2.13 all the phones work great with NTP and with 1.1.0.1 all the phones don't work so I dont think its an actual phone problem or a network problem since reverting to 1.0.2.13 have the problem. Also if you read in the comments at the bottom of the page, other people have the same problem. Maybe only newer phones are affected? Mine are all pretty new. - mike240se
    • NOTE: (Mar29/06) Make sure you don't have DHCP option 42 overriding the phone's NTP settings. - bani
    • NOTE: (Mar30/06) I triple checked and tried everything, it just keeps going back to January 2nd after every reboot, I downgrade the firmware and NTP works perfect. - mike240se
    • => Workaround: (APR02/06) I discovered what the real bug is here (I think) :). I'm getting the same thing after changing the daylight savings time option to yes. What I needed to do was select Daylight Savings to no, and change my timezone instead (to fake the right time). It has to be a bug in the Daylight Savings option. - Mike
    • NOTE: (Apr02/06) I can't reproduce, I've selected my time zone and daylight savings works as expected. - job
    • NOTE: (Apr07/06) I also can't reproduce this NTP problem, and setting daylight savings settings makes no difference to me... - SoloFlyer
    • NOTE: (Apr07/06) You guys fixed... err, jinxed me: now I can't reproduce it anymore either. Maybe it was a momentary glitch with my network or phone connecting to the NTP server. Anywho, I'm glad its working. Should we mark this as a non-issue now, or is anyone still experiencing this problem? - Mike

  • MINOR: (APR17/06) When turning on the account option: "Unregister on reboot" the phone reregisters every 30 seconds regardless of timeout settings. With 1.0.2.8 this was also pressent but the reregistrations occured every 60 seconds. When "Unregister on reboot" is set to no the phone works like it should. - Falle

  • TWEAK: (APR05/06) Time Display while on Call: I would like to see the time displayed while on a call in the upper right corner of the display, regardless of off-call preference. More often than not I'll find myself making appointments or what-not and will be looking for the time (and not the date, since I have it set to show time in large when not on a call. - Mike

  • TWEAK: (APR17/06) When a BLF is lit(remote user is on the phone) the phone silently ignores when the user presses the corresponding button. What If the user wants to leave a voicemail message for this person or maybe show up as "Call waiting"? If there must be such a block it should be possible to disable it in the configuration. - Falle



Firmware Notes (Beta 1.0.2.13):


This is an internal build correcting issues in the current Beta. Distribution of this build on this Wiki has been sanctioned by Grandstream. - thetatag

The firmware 1.0.2.13 is attached to this page:
Image


Changelog
Build 1.0.2.13 2/21/2006
  • Fixed LCD problem on older units
  • Fixed keypad not responding if boot up without network cable
  • Fixed when LINE keys are pressed when in GUI MENU, the usual "Dial Using" prompt is not displayed
  • Added support for DHCP option 2 (Time Offset). Provision variable P143, possible values 0/1. When set to 1, it will override the configured Time Zone setting if available. Default is No (0).
  • Added support for DHCP option 42 (NTP server). Provision variable P144, possible values 0/1. When set to 1, it will override the configured NTP server. Default is No (0).
  • Added support for DHCP option 66 (TFTP server). Provision variable P145, possible values 0/1. When set to 1, it will override the configured provision path and method. Default is No (0).
  • Added configurable DHCP options 12 (host name/P146), 15 (domain name/P147), and 60 (Vendor Class ID/P148). Max length allowed is 32 bytes each.
  • Fixed we do not follow the port in the Record-Route URI if there is maddr attribute
  • Added we will attempt to start the initial provisioning routine every minute for 5 times unless we received any response from the server (any type of HTTP error or OK response, and TFTP error or data response will stop further trying)
  • Changed we will restart STUN accounts when STUN port mapping changed (previously we only check IP address change). Also changed: STUN checking interval is now the same as the configured keep-alive interval (used to be 1 minute fixed)
  • Changed the RTP keep-alive interval (when a call is placed on hold) to the same as the configured keep-alive interval (used to be 25 seconds fixed)
  • Fixed we do not send RTP keep-alive interval during MUTE
  • Fixed when early-dial is in use, the phone times out 4 seconds after the last incomplete digit is dialed instead of the configured no-key time out value configured. Also fixed deletion of an incorrect timer due to this action.

  • Fixed a string matching function which made our password checking case-insensitive
  • Fixed VLAN TCP issue
  • Added model number and firmware version (app) in every syslog message
  • Changed- In GUI Status menu item, "N/A" is displayed instead of "NO" for accounts configured to be not-in-use
  • Fixed the 3WC related problems by application restricting 1PCM+1LBR conference rule
  • Fixed under "Ring Volume" the volume is decreased by 1 each time you enter that menu item even if you did not make any change
  • Fixed we do not detect duplicate firmware (and fall into reboot loop) if the provisioning server responds first data frame < 512 bytes
  • Fixed we crash after receiving fragmented SIP messages introduced in 1.0.2.5

Bugs / Tweaks:

Please sign your posts and also enter a comment for the wiki history! Pages of unknown edits are not helpful!
  • MAJOR: (Mar31/06) Phone crashing The Phone crashes reproducable when more than eleven concurrent calls are sent to one account on the GXP2000 - Mitsch
  • MAJOR: (Apr18/06) Phone locked up The phone locked up when I was scanning through Missed Calls looking for a way to add the number to the phonebook - ADW
  • MAJOR: (Feb23/06) Rebooting. The phone seems to do a complete power cycle, including the ethernet switch, every 15-20 minutes. This has not happened in a call, but I don't see why it couldn't. - russlevy
    • NOTE: (Feb24/06) Check your ADVANCED SETTINGS -> Firmware Upgrade and Provisioning -> Automatic Upgrade settings. - bani
      • NOTE: (Feb25/06) I have already downgraded back to 1.0.2.8, since it was unusable in its other state. Automatic Upgrade was set to No, so that couldn't be the problem. - russlevy
        • NOTE: (Feb25/06) Do you have an old hardware revision? - mike240se
          • NOTE: (Feb26/06) No again :) - russlevy
    • NOTE: (Mar29/06) I have experienced reboots after 12-15 minutes while in calls every time, and it is very frustrating. It happened three times in the same conference call. I have never noticed the phone reboot while idle. This is major, since I'm trying to show a cheap imported phone will work as well as a Cisco 7960

  • MAJOR: (Feb24/06) Blanking Screen. Since installing 1.0.2.13, my phone's LCD becomes blank after about 10 seconds after a reboot, and the only way to fix it is to reboot. I've seen my screen display information for maybe 60 seconds total since I upgraded. - NateBell
    • NOTE: (Feb24/06) I can confirm screen blanking still occurs on phones with MAC addresses starting with 00.0B.82.03. The screens blank almost immediatly after reboot. Firmware seems ok on all other MAC addresses. I have this loaded on about 20 phones now testing. 5 of them are the older phones with MAC's that start with 00.0B.82.03. - phoneguy30
    • NOTE: (Feb25/06) I can also confirm its still affecting my old-MAC phone ... it goes blank after anything from 30 seconds to half an hour. once blank, it stays that way, it might still be corrupted , but hard to tell once its blank ;) . - robin_sz
    • NOTE: (Feb25/06) I'm not getting this on my old-MAC phone. The whole bisplay blank/scroll/corrupt has gone away on this beta, so far. However, it should be noted that I have the missed call list turned OFF. - jedi98
    • NOTE: (Feb27/06) Both my 00.0B.82.03.xx.xx test phones are still having the blanking problem, though it can be anyware from several seconds up to several hours before the screen goes blank... - SoloFlyer
    • NOTE: (Feb28/06) Screen blanking is not related tothe missed call list as i have this turned off on one of my test phones and it still has the problem - SoloFlyer
    • NOTE: (Mar11/06) Screen blanking has now cropped up again on my phone but was NOT triggered by an incoming call. The phone re-boots @ 6:30am (with a cron job), display was blank by 12:00 midday. This has happened 3 times so far, approx once every 3 days. - jedi98
    • NOTE: (Mar23/06) Time drags on .. still no update and my phone si still useless with a blank display ... any chance of grandstream offering swap-outs to those with the "problem" hardware for a newer phone? - robin_sz
    • NOTE: (Mar23/06) Keep in mind the entire 1.2 series is still technically a flash-at-your-own-risk beta. The first 1.2.x firmware didn't have the display bug as i recall, maybe use that for a while? I think GS is still crunching on the feedback they got from .13. Soon hopefully :) - Helix

  • MAJOR: (Feb24/06) Missed Call Lockups. The phone will lock up under some conditions while viewing and deleting items in the missed-call list. It appears that a power-cycle is required to get the phone back into a functional state. While I've observed this on multiple phones and multiple times on each, I have not found a guaranteed way of reproducing the bug. - thetatag

  • MAJOR: (Mar06/06) Random Lockups. Aside from those lockups described above under Missed Call Lockups, several GXP-2000's that I have tested will randomly lock up for no aparent reason, sometimes during a call, and sometimes while sitting idle. This is seperate from the rebooting / power-cycling issues discussed above, as the phone gives no indication that it has locked up, except that the clock stops advancing (nothing changes on the display) and sometimes the MWI light remains lit. In my test cases, this happens infrequently enough to not completely inhibit production usage, but often enough to be annoying — maybe every two to four days. - thetatag
    • NOTE: (Mar07/06) Are you able to ping the phone? Send a curl reboot? Are you using any special options? 1.0.2.13 i assume? - mike240se
    • NOTE: (Mar10/06) This may not be the same crash, but it sounds similar, though we get a crash maybe once or twice a month. We have a bunch of phones running 1.0.1.9 and every now and then one of the phones locks up. The phone can't be pinged, you can't access the phone's web interface. Hard crash. The MWI light turns on, and the display freezes. I noticed this tended to happen when a person was on the phone, and another person called the phone. When the caller got to the voicemail, the phone locked up. I turned off call waiting to see if that would help, and so far so good. Of course, this was with older firmware, so I don't know if the issue has been resolved in 1.0.2.13. -NateBell

  • MAJOR: (Feb06/06) qualify=yes in busy environments will sometimes cause 1-way audio when calls are resumed after being put on hold with MOH. (see asterisk-users post). a quick fix is to qualify=no and restart asterisk. - bani
    • NOTE: (Feb14/06) I have experienced similar condition with qualify = yes. When calls are resumed from MOH, caller is unable to hear our side. This only happens when there are many GXPs in use at the same time. I cannot reproduce condition unless our phones are active — cannot duplicate with just 1 SIP client/user. The only way I have been able to fix it is make qualify = no, and restart asterisk. - SamL
    • NOTE: (Mar23/06) I've just experienced the same problem on firmware 1.0.2.13 also It's only happening on 1 remote user going over multiple NAT's the other 10 with the same conditions are fine. - ChrisUK

  • MINOR: (Apr04/06) MWI doesn't work with STUN The Message Waiting Indicator doesn't seem to work with STUN - but does work with Outbound Proxy - crayfishuk

  • MINOR: (Mar28/06) Numbers off-screen. When entering long phone numbers (i.e. for international calls) the phone number goes off-screen as it advances, rather than moving up to the upper line as it used to. - Mike

  • MINOR: (Mar27/06) No Dialtone. Normally pressing a line button causes a speakerphone with dialtone. Today, pressing a line button gave no dialtone. In fact all dialtones had disappeared but the phone still worked. Rebooting fixed this & it has only happened once, hence it's a minor. - Jedi98

  • MINOR: (Mar21/06) Silent Ringing. When you call another SIP phone you don't get the "Ring-Ring" and is completly silent until the remote party picks up at which point it works as normal. Its completly random and happens often enough for me to be worried about putting it into a production environment. Any Ideas?. - ChrisUK
    • NOTE: (Mar23/06)If you're using Asterisk, add progressinband=yes in sip.conf, and you will allways hear the ringing. - mtryfoss

  • MINOR: (Mar13/06) Asterisk BLF. If the SIP account's username begins with "id", the BLF doesn't work: the phone subscribes, but BLF indicators don't light. It seems that it was not the case with the previous beta version. - snipefoo

  • MINOR: (Mar13/06) Voice Hic-ups. In the 1.0.2.13 firmware there is a 1 second hic-up / stutter at the start of the call when using full volume with PCMA and asterisk. This happened once with the first call after the firmware was first installed and has happened 2-3 times in a 4 day period and apears to be random. - ChrisUK

  • MINOR: (Feb27/06) Display Corruption - Artifacts. In the 00.0B.82.03.xx.xx phones there still appears to be a small amount of corruption left in the screen specifically in some of the decimal places , this was happeninging in the 1.0.2.8 firmware also but it is not evident in the photos that i had taken as the quality wasnt good enough.- SoloFlyer

  • MINOR: (Feb22/06) Display Issues. The 484 message when a time-out occurs partway through a number in early dial has been observed at least once as being displayed as "484ÿ". Grandstream has observed “503EE" on the display. This suggests that there may be a NULL termination issue or a pointer issue. Alternatively, there may be an issue with clearling the display buffer. However, reliably reproducing this minor bug has proven to be difficult. Please post an additional note if you discover a way to force this bug to manifest itself reliably.- thetatag
    • NOTE: (Feb22/06) easy to reproduce. exten => 5555,1,Congestion will do it. Apparently a response of SIP/2.0 503 Service Unavailable causes the gxp2000 to print 503EF - bani

  • MINOR: (Feb24/06) Early Dial / Transfer. When using the handset (not speakerphone) and early dial, after completing a blind transfer, the GXP-2000 goes back to a dial-tone and suggests that you can dial out again. However, the buffer that stores the early-dial number is not cleared, so the first digit you dial will be sent with the transfer number prepended to it. This is true even if you press another line button before dialing. (However, if you hang up the handset and pick it up again, the dialing begins fresh. Grandstream has acknowledged this as a bug a few versions ago, but has not yet corrected it. - thetatag

  • MINOR: (Feb24/06) Muted DTMF. You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag

  • MINOR: (Feb24/06) Paging Rings. Calling the phone with the SIP header "Call-Info: answer-after=0" will not cause the phone to answer immediately. The phone produces about half of a ring and then answers. At 0 there should be no ring, at 1 there should be one ring, etc. Grandstream has acknowledged this as a bug several versions ago, but has not yet corrected it. - thetatag
    • NOTE: (Feb27/06) answering after 0 rings would cause the recipent to have no indication that anything had happened, allowing for abuse... I suspect that this is not a bug but a deliberate feature ( or its a bug which they have chosen not to correct for the aforementioned reason ) a true answer after 0 rings should be added as a configuration option but it should not be enabled by default - SoloFlyer
      • NOTE: (Mar1/06) I agree that the half ring is very nice to have, if they do fix this so called "bug" i would also like to see the option to turn on the half ring. Without, it could be used to secretly monitor peoples conversations. - mike240se

  • MINOR: (Feb24/06) Asterisk BLF. If Account "SIP Registration" is "no" (ie. sip.conf uses host=xx.xx.xx.xx, not host=dynamic) BLF does not subscribe. - lonnie

  • MINOR: (Feb14/06) gxp2000 gsm codec is broken with asterisk music on hold. ulaw and g729 work fine, only gsm is broken. - bani
    • NOTE: (Feb17/06) same here - i think this has to do with silence suppression - when using GSM, silence suppression is obiously on (or it appears to be) even though the feature is turned off in the options - Adrian2k
    • NOTE: (Feb27/06) I had a call today using gsm and could not use the transfer button or the conference button. When trying to recreate it happened everytime. Anyone else haveing this issue? - hminc

  • MINOR: (Feb27/06) Ring volume configuration doesn't survive to a reboot. - snipefoo

  • TWEAK: (Feb22/06) Do Not Disturb. There is no reason for the Do Not Disturb menu to offer enabling and disabling DND mode. One of the options is already the current state. Only the other state and "Cancel" should be offered. Perhaps it should also tell you the current state (a dot next to the current mode, almost check-box style). There are also far too many button presses required to turn DND on and off since the GUI menu was introduced. - thetatag
    • NOTE: (Feb22/06) Probably a better way would be to just use MUTE/DEL as a DND toggle on the idle screen. - bani
    • NOTE: (Feb23/06) Now the mute/dell makes loads of sense because the DND is something you just want to press & go (out for lunch, coffee, snooze etc...) also it's intuative. - jedi98

  • TWEAK: (Feb24/06) Speakerphone volume setting should be independent of handset volume. I need to turn the speakerphone up to full volume to hear it, but this means the handset is too loud. - bani

  • TWEAK: (Feb24/06) Turn off speaker on remote disconnect should default to Yes - bani

  • TWEAK: (Feb25/06) Missed Calls. Its still painfully difficult to get rid of the missed calls notification. Hitting the up arrow should be enough to remove the notification. Or atleast make it a choice in the web gui. Having to either look at every call or scroll all the way to the bottom of the list to clear the list is too much work, especially if there are like 75 calls on the list. Yes you can turn off missed calls completely, but id rather have them. - mike240se

  • TWEAK: (Mar06/06) BLF / Speed Dial. When you press a non-BLF speed-dial button without being on a line, the speakerphone engages and the call progresses. This is appropriate behavior. However, if the speed-dial button is set to BLF, you can still call it, but you have to pick up a line first. It will not automatically engage the speakerphone. It seems to me that the BLF buttons should behave exactly the same as the speed-dial-only buttons, except with added BLF indication. (I have repeatedly asked Grandstream to change the drop-down selection of "Speed Dial" or "Asterisk BLF" to a checkbox to enable or disable BLF, so as not to suggest that it loses its speed-dial capabilities when using BLF.) - thetatag
    • NOTE: (Mar10/06) When the phone is IDLE and I press a speed-dial or a BLF button, the speakerphone engages (is it due to a particular setting ?). I think it would help some users if the speed-dial/BLF buttons were able to take a new line and place the active line 'On Hold', when you're already on line (to make easier the call transfer process) - snipefoo

  • TWEAK: (Mar18/06) Inverted colors Why the heck is the menu "inverted"... The whole office here agrees, that this is plain ugly... Blue on White with a more appealing slider would make it look far more professional... IMHO - KampfCaspar

  • TWEAK: (Mar18/06) BLF pickup in dialled numbers Picking up a BLF key effectively dials "**[extension number]". This process should NOT be listed in "dialled numbers" as no user interprets it as dialling anything. - KampfCaspar

(In addition to adding user feedback here, please move items from the 1.0.2.8 bug/tweak list to this list if you discover that they are still issues! Thank you! - thetatag)





Firmware Notes (Beta 1.0.2.8):


The firmware 1.0.2.8 is also attached to this page:
Image

The template for the GXP2000(w/ updated firmware) is here : http://aseclub.net/bigb/gxp2000/gxp2000_config_1 0 2 3.txt

Changelog
Build 1.0.2.8 2/6/2006

  • Fixed memory error syslogs not including right source

Build 1.0.2.7 2/3/2006

  • Fixed under "Ring Volume" the volume is decreased by 1 each time you enter that menu item even if you did not make any change
  • Fixed we do not detect duplicate firmware (and fall into reboot loop) if the provisioning server responds first data frame < 512 bytes
  • Fixed we crash after receiving fragmented SIP messages introduced in 1.0.2.5

Bugs / Tweaks:


  • MAJOR: Display Corruption Blanking, Wrapping, Upside down, and Artifacts... - SoloFlyer
    • NOTE: (Feb??/06) Display blanking problem from 1.0.2.3 still present where the display will clear completely until re-boot following an incoming (missed?) call. Display stays lit but blank. Also has been triggered by calling loopback call to the phone. This happens most but not every every call. There is a delay between the incoming call and the clear screen. - jedi98
    • NOTE: (Feb??/06) When using Speakerphone, display will often wrap so the top line (status icons/clock) are on the bottom. Due to some other oddness and that the divider below them is still there at the top, I'm assuming this is a bug. - Helix
    • NOTE: (Feb06/06) I can't reproduce this. Can you attach a screenshot of the bug? - bani
    • NOTE: (Feb07/06) Upon further research it's not a sp-phone bug, it just got triggered when I hit SPKR. This is the same bug discussed above that made beta pull this ver off his TFTP, my MAC addy is 00.0B.82.03.xx.xx so I am affected. See: here if you still want a screenshot. -Helix
    • NOTE: (Feb07/06) I can confirm this backward text bug still exists in version 1.0.2.8. I am using a phone with a MAC starting with 00.0B.82.03 as well. -Flu
    • NOTE: (Feb16/06) I can confirm, Screen Wrapping, Screen Artifiacts, Screen Blanking, and Screen upside down... Both the phones i am testing with have 00.0B.82.03.xx.xx MAC addresses.... Images: Upsidedown, Upsidedown and Corrupted Upsidedown and wraped, Wraped - SoloFlyer

  • MAJOR: (Feb06/06) qualify=yes in busy environments will sometimes cause 1-way audio when calls are resumed after being put on hold with MOH. (see asterisk-users post). a quick fix is to qualify=no and restart asterisk. - bani
    • NOTE: (Feb14/06) I have experienced similar condition with qualify = yes. When calls are resumed from MOH, caller is unable to hear our side. This only happens when there are many GXPs in use at the same time. I cannot reproduce condition unless our phones are active — cannot duplicate with just 1 SIP client/user. The only way I have been able to fix it is make qualify = no, and restart asterisk. - SamL
    • NOTE: (Mar23/06) I've just experienced the same problem on firmware 1.0.2.13 also It's only happening on 1 remote user going over multiple NAT's the other 10 with the same conditions are fine. - ChrisUK

  • MINOR: The ring volume setting doesn't survive to a reboot. -Adrian2k

  • MINOR: Loopback call (dial phone from itself, via asterisk) drops out on answer (expected) but the display shows missed call while whe phone still thinks it's off-hook (speakerphone) so up/down arrow adjust volume instead of missed call list. -Jedi98

  • MINOR: The ringtone selection (via display menu, but not via Web GUI) need a reboot to operate (that wasn't the case in the 1.0.1.x versions). - Snipefoo

  • MINOR: (Feb11/06) In Early Dial mode, the Delete button doesn't delete the last pressed digit from the screen. To reproduce, set phone to early dial mode, start dialing a number, then press delete. The deleted number(s) won't go away until a new number is pressed. - Ted

  • MINOR: You still cannot send DTMF digits while the phone is on MUTE. You hear them, but they don't get sent. - thetatag

  • MINOR: (Feb14/06) Phone displays 503EF when it should display Busy instead. - bani

  • MINOR: (Feb14/06) While phone is ringing, if you press UP, DOWN, or MENU key and then exit menu, phone rings with blank display. - bani
    • NOTE: (Feb20/06) Its not actually blank it still shows the bar at the top, but that is all it shows it goes back to normal after the phone stops ringing though... -SoloFlyer

  • MINOR: (Feb14/06) gxp2000 gsm codec is broken with asterisk music on hold. ulaw and g729 work fine, only gsm is broken. - bani
    • NOTE: (Feb17/06) same here - i think this has to do with silence suppression - when using GSM, silence suppression is obiously on (or it appears to be) even though the feature is turned off in the options - Adrian2k

  • MINOR: (Feb17/06) BLF lights stay on after the phone looses the connection to the Asterisk box - Adrian2k

  • MINOR: (Feb19/06) I have 2 registered accounts on Line 1 and Line 2. Line 3 has wrong settings, so it can not register and shows "Not Registered" when the phone is lifted up. When I press "LINE 2" and back to "LINE 3" the message "Not Registered" disappeared. In reality, it's still not registered. - PaulK

  • MINOR: (Feb21/06) When the phone has no ip assigned (network unplugged) you can't access the phones menu system, hitting the round key does nothing. For example, i needed to change the registration config so it wouldnt conflict with a phone already on the network, but couldnt. - Mike240se

  • MINOR: (Feb22/06) I phoned someone over the phonebook and didn't lift up the handset. When he hang up the phone my GXP2000 began beeping the busy-noise for a whole minute (or more?). It would be better to stop the busy-noise after 3 seconds. - PaulK
    • NOTE: (Feb24/06) I don't think this is a bug. Try Account 1-4 -> Turn off speaker on remote disconnect -> Yes. If this fixes the problem for you, you should remove this from the buglist. - bani
    • NOTE: (Mar05/06) If that is the case, then yes should be checked by default (i just noticed, you stated it already above). Why would you want to hear the busy-tone all the time? Anyways, I gonna check if that fixes it. - PaulK

  • TWEAK: (Feb19/06) (issue from 1.0.2.3) There's still a spelling error somewhere: "not alloweded"- PaulK

  • TWEAK: (Feb19/06) (issue from 1.0.2.3) It should be enough to see into the "Missed Calls" list to let the "1 Missed Calls Press /\ key to ..." frontal remindermessage disappear.- PaulK
    • NOTE:(Feb21/06) I agree this is extremely annoying, there should atleast be a setting in the web interface where you can decide how you want this handled. - mike240se

  • TWEAK: (Feb13/06) On Missed Calls list, RIGHT button should jump to Clear All, and UP button at the top of the list should jump to the bottom of the list.- bani
    • NOTE: (Feb16/06) Fundamentally, it seems, the main problem with missed call is there's no "new missed calls" concept. How about del key clears the screen message so we can see the real screen & turn the light off? Config option to NOT show screen message because it's a pain in ringing groups of phones. - Jedi98

  • TWEAK: Speakerphone volume setting should be independent of handset volume. I need to turn the speakerphone up to full volume to hear it, but this means the handset is way too loud. - bani

  • TWEAK: (Feb15/06) Config->Factory Reset is too cumbersome to use. 2 or 3 levels of "are you sure?" would be sufficient, but requiring the entire MAC address is way over the top — that is like 12 levels of "are you sure?". This makes it incredibly cumbersome when you need to factory reset dozens of phones. - bani
    • NOTE (Feb21/06) There should be a way to factory reset when plugging in the phone, like plug in the phone while holding 2 keys down at the same time. - mike240se





Firmware Notes (GXP-2000 firmware v1.0.2.6)


Changelog
Build 1.0.2.6 2/2/2006

  • Fixed GUI sometimes send additional key events (cause of the SIP configuration problem). Now the focus is default at the "CANCEL" button in each SIP edit dialogs. This also resolves the Upgrade menu inaccessible problem
  • Fixed GXP displays "TFTP Provisioning" briefly before correcting it to "HTTP Provisioning" if HTTP is in use
  • Fixed ring tone file problems
  • AGC change
  • Changed- FUNC keys would act as speed dials even when BLF status is BUSY
  • Changed- All keys are blocked when provisioning is in progress (including LINE keys and SPEAKER key)
  • Changed- Factory reset will now clear phone book as well as custom ring tones
  • Changed- GUI->Config->Upgrade changed to allow edit firmware and config server (the original interface was implemented before the application TFTP), Note that the old 12 digit IP address format you use for direct IP calling is no longer valid here. You will need to type in the dots (* key) to separate the octets.
  • Fixed we fall into reboot loop when there exists a ring3.bin in the server and ring2.bin spans over 64k
  • Fixed we do not reboot immediately upon receiving a cfg file (that caused a change)
  • Changed- we no longer reboot if only ringx.bin are downloaded
  • Fixed we display line status as MUTE if the previous call is in MUTE and disconnected by remote party
  • Fixed when account name length > 16 characters, GXP-2000 freezes after a call (Bugzilla #135).
  • Added allow user to use the Speed Dial keys to do blind transfer
  • Fixed we do not process the IP packets if the first fragment did not arrive first
  • Fixed we still display AM/PM even when 24 hour display mode is selected
  • Fixed our LCD backlight does not light up immediately when a call comes in
  • Added- LCD backlight stays light up when there are unviewed missed calls to alert user, the LCD backlight also stays up whenever the MENU operation is in progress
  • Fixed GUI SIP configurations were not accessible
  • Changed- GUI menu item sequence rearranged for user-accessibility, Status page items rearranged for a quick glance at registration status
  • Changed- when display mode is set to DDMMYYYY and reverse date/time is set to YES use period as separator instead of hyphen (I have seen many Europeans request for this and this does make sense so that when you see a date 03.04.2006 you can tell if 03 is the month or 04)
  • Changed- when display mode is set to DDMMYYYY and reverse date/time is set to NO, the long date string is displayed as "dddd, dd MMMM" where "dddd" is day-of-week in English, "dd" is day-of-month in number, and "MMMM" is month in English, example: "Friday, 27 January" (standard mode is "dddd, MMMM dd")
  • Added allow disable miss-call features as per-account setting, changing this setting takes immediate effect without reboot. Incoming call is still logged, only missed calls are not. It can be provisioned using P182/442/542/642, valid values 0 and 1, default is 0 (No) which WILL log all missed calls.
  • Added disallow MENU actions when provisioning is in progress
  • Changed-when you use the UP arrow key to view missed calls, you will return to main idle screen directly if you either use the LEFT key or delete all missed calls
  • Changes to phonebook: on the main Phone Book menu, the "Add" is renamed to "New Entry", in the New Entry page the "Add" is renamed to "Confirm Add", "Back" is renamed to "Cancel & Return"
  • Fixed phonebook entries remains in flash (and reappear after reboot) after delete

Bugs / Concerns:


  • MAJOR: Endless http upgrade reboot bug is still there. Seems the GXP2000 is confused by the http responses returned by Apache 2.0.53/54 on Fedora Core 4. Anyone can reproduce the bug by pointing your phone's Firmware Server Path to mihoshi.anime.net/gsexperiment at your own risk! do not do this unless you are able to recover from an endless upgrade reboot cycle on your own. i will not be around to help you recover your phone if you cannot. - bani
BUG FIXED IN 1.0.2.7 ALPHA. NEXT BETA RELEASE SHOULD NOT HAVE THIS BUG. - thetatag

  • MINOR: Everytime you go into the ring volume setup, the volume decrease of one unit. - Snipefoo
BUG FIXED IN 1.0.2.7 ALPHA. NEXT BETA RELEASE SHOULD NOT HAVE THIS BUG. - thetatag




Firmware Notes (GXP-2000 firmware v1.0.2.3)


As of 1/26/2006, Grandstream has officially released for Beta testing GXP-2000 firmware version 1.0.2.3. Following the very successful format of detailing the last firmware beta, the following lists features implements, bugs, etc. For complete release details, see Release Notes. - thetatag

GXP-2000 firmware is larger than other Grandstream product firmware and downloading time is normally 15 minutes.
Also, there are major changes in firmware 1.0.2.3 which requires firmware to be downloaded twice and power cycle as well
- DigiGuru

WARNING:
(Feb01/06) Grandstream has stated that there is currently no way to downgrade from the 1.0.2.x to the 1.0.1.x series. If you upgrade to the 1.0.2.x firmware, you are stuck with it. Grandstream has indicated that a transition file may be available at some point which would allow such a downgrade of firmware, but it is not available now. If you choose to participate in the beta, you have been warned! (Fortunately, this version seems pretty good for the most part, so I doubt too many people would regret moving in this direction, especially if you have a non-production environment in which to test it.) - thetatag

Changelog
Build 1.0.2.3 1/24/2006

  • Added GUI Interface
  • Added Address Book
  • Added Call Log
  • Shortened the wait-time between downloading files from 1 second to 100ms
  • Added configurable T2 interval--this is a per-account setting. Possible values are 2/4/8 sec. Provision parameter P250/P441/541/641, valid values 200/400/800 (in 10ms unit). Invalid values ignored, default value 4 second.
  • Fixed display fonts not aligned correctly during call-time (the call-timer ticker should align to the right of screen)
  • Fixed miss call count does not display the 0 at the end
  • Changed max call log to 50 entries
  • Added configurable T1 timeout interval--this is a per-account setting. Possible values are 0.5sec/1sec/2sec. Provision parameter P209 (P440/540/640 for accounts 2-4 on GXP-2000), valid values 50/100/200 (in 10ms unit). Invalid values ignored, default value 1 second.
  • Added under Broadsoft mode, DNS SRV fail-over happens after 3 retries (so if you set T1=0.5 sec, it takes 7.5 seconds to fail-over to second server).
  • Fixed session-timer does not work properly before session establishes (UPDATE/481 issue reported)
  • Fixed we do use increment CSeq in the INVITE to unhold--this is a general 100rel bug what will occur whenever 100rel (PRACK) is enabled and impacts GXP-2000 as well.
  • Added LCD displays provisioning status and warning message when flash writing/erasing is in progress
  • Added provisioning protection- During provisioning all incoming SIP packets will be dropped without processing
  • Added implementation of T2 timer (see RFC3261) and send BYE when 200 OK time-out
  • Added we will do TFTP/HTTP provision upon DHCP/PPPoE completion if no IP address was available initially
  • Fixed VLAN bug
  • Added TCP/HTTP fix
  • Fixed TFTP retry issue
  • Fixed we respond 200 OK with "event: presencenoevent" for the SUBSCRIBE (event: presence) we receive
  • Fixed when reversed-date/time selected and DD-MM-YYYY is selected, the date display is not displayed properly

Features / Fixes

  • FEATURE: Menu. When the phone is on-hook, the round button brings up a simple text menu (complete with scrollbars) giving access to various configuration options. It is much more useful than the old-style configuration and gives that much more credence to the GXP-2000 as an enterprise-grade phone. - thetatag For unknown reasons, the menu system uses inverted colors which looks strange. Most functions now requires more key presses to access. No menu selection using the keypad. Uses OK and Cancel buttons for actions which has no corresponding keys, you have to access them by combinations of round key and arrow keys (depending on where cursor is positioned, which isn't obvious), making it hard to navigate. - job
  • FEATURE: Phone Book. The phone book is only partially useful at this point. You can manually add entries to it and use it to dial those entries, but there is currently not support for grabbing a number from the call logs (either incoming or outgoing) and placing them into the phone book. So while it might be slightly useful, be aware that you'll have to enter everything manually, and it doesn't appear that you can use configuration files to include complete phone books in your provisioning. Also note that you cannot currently assign phone book entries to any sort of speed dialing. They stand alone. You go into the phone book if you want to use it. Also, it does not use the name associated with a number in your phone book for CallerID. - thetatag
  • FEATURE: Missed Call List. When the phone rings and you don't answer it, the screen displays, for example, "1 New Missed Call" "Press ^ Key to view". Pressing the up arrow brings up a list of missed calls, including time/date stamping, account number, and callerID information. It also gives you the opportunity to redial one of these entries. Currently there is no way to enable or disable this feature on an account-by-account basis, so if you use a seperate account for all of your internal calls (for distinctive ringing or whatnot), you cannot keep these numbers from showing up in the list even if you don't care to track missed internal calls. - thetatag
  • FEATURE: Do Not Disturb Icon. The "Do Not Disturb" mode now has a nice flashing icon at the top of the display, rather than announcing itself in place of your time or date. Regretably, there is still no account-by-account DND mode. In addition, the DND option is now under "Preferences" in the menu, requiring several more button presses to turn it on and off. - thetatag
  • FEATURE: Paging Support. If you pick up a line (so the phone says "DIAL USING") and press the round button before dialing, the prompt changes to "PAGE USING". This simply inserts the SIP header for auto-answer support. If you're doing anything other than peer-to-peer (which doesn't support the page option, to my knowledge), your PBX will need to look for that SIP header to do anything useful with it. - thetatag
  • FEATURE: NAT/Router Support. This firmware now allows the GXP-2000 to act as a Network Address Translation router, with the LAN port being the "outside" and the PC port connecting to the internal network. Through the web interface the GXP-2000 can be configured for NAT or traditional bridging (the behavior of prior versions). There is currently no support for non-NAT routing. - thetatag
  • MINOR BUG FIX: The problem in 1.0.1.13 in which all codecs did not appear for every account has been corrected. There are currently 5 supported codecs (a disappointment for those of us hoping to see SPEEX): GSM, G.723.1, G.729A/B, PCMU, and PCMA. Why there are eight options in the preference list is still a mystery. - thetatag
  • MINOR BUG FIX: With the new menu system, you no longer have to pick up a line to use the recent callers and recently called lists. In fact, just the opposite is true. There is no longer any way to access these lists while off-hook. - thetatag

Bugs / Concerns

  • MAJOR: No longer possible to do conference calls with latest beta firmware. - Michael
    • NOTE: (Feb01/06) Michael: I have tested this on my GXP-2000s and have not had any loss of functionality. (I'm not willing to commit to it working perfectly even on mine!) Could you explain in a little more detail exactly what the phone does or doesn't do when you try to use the conference function? - thetatag
    • NOTE:(Feb3/06) I have just set up my third GXP-2000.with the latest beta firmware. Conferencing works perfectly on all three phones. - emckinnon
  • MAJOR: Both tested phones died (bridged network). Soft reset doesnt bring network back, only hard restart solved this issue (no dhcp, static IP, network LEDs do nothing). - Festr
  • MAJOR: Lines can't be called when BLF status is busy - most of my extensions are capable of muliple simultanious connectons. - Adrian2k
  • MAJOR: The phone goes into an endless loop with firmware upgrades on this version. After the phone reboots for the second time (after fetching boot55a.bin and gxp2000a.bin) you need to rename the files so the gxp can't grab them again. Otherwise it will just keep grabbing the same firmware over and over again. - bani
    • NOTE: (Jan27/06) Bani: I have been in touch with Grandstream concerning this bug report. Please provide some additional information. What firmware version were you upgrading from? Which upgrade method did you use? Did you completely replace all of the files for the upgrade (including boot55.bin)? Are you using any custom ringtones? If so, how many and what are their sizes? Please verify that none are larger than 64K as well and that none are the old BT100 ring tones. I have upgraded several phones without being able to recreate this problem, so any additional information would help diagnose and correct it. You can contact me at admin {at} composite-tech {dot} com. Thank you. - thetatag
      • NOTE: (Jan30/06) thetatag: happens via http updates. no custom ringtones at all. i am 100% certain the bin files are correct. other users on #asterisk have been able to reproduce the problem. - bani
    • NOTE:(Feb3/06) The first two GXP-2000 phones that I upgraded via tftp had this problem. I believe it relates to BT102 stock ringtones in the tftp directory. I renamed the BT102 ringtone files and added three custom ringtone files created with the tools available from the Grandstream web site. The third upgrade via tftp succeeded the first time. The file, ring3.bin does not load. Files ring1.bin and ring2.bin do. Its not the ring3.bin file as swapping 2 and 3 allows the former ring3.bin to load. - emckinnon
  • MAJOR: The display went blank completely after a period of about one day and would not show anything until a re-boot. After another day running (idle) the display blanked again. Phone previously had v1.0.1.13 firmware. - jedi98
    • NOTE: (Jan27/06) Since the beta has only been out for a day and a half, obviously it's too soon to tell if this is a freak incident (static electricity, swamp gas, sun-spots, etc.) or a major firmware issue. But if it is a firmware problem, it's definately worth making Grandstream aware of. Were you using the phone when this happened? Has it happened to multiple phones? Also, if anyone else reading this has had this problem, please indicate such and provide some basic information on any phone-related events leading up to the crash. I will present any information this community can put together to my technical contact at Grandstream. Thank you. - thetatag
    • NOTE:(Jan27/06) I have encountered this bug as well. I am testing the new firmware on 2 phones right now, and this morning when I got to work, one had a blank screen. It still seemed to be functional, but I could only restore the screen by reseting the phone. The other phone is doing fine. I don't know what the conditions were when the screen blanked. All I know is it happened over night, so I doubt anything out of the ordinary occured. I'll update if it happens again over the weekend. NateBell
    • NOTE:(Jan30/06) This bug same for me, tested two phones . Festr
    • NOTE:(Jan30/06) Over the weekend the screen on the same phone as before turned off. Curiously, the other phone I'm testing with does not have this bug. The only difference between these phones (besides differing SIP numbers) is the format for the date/time. The phone with the blank screen had the time displayed in the center while the other has the date in the center. Both phones still seem to operate fine. NateBell
    • NOTE:(Feb3/06) Two of my GXP-2000 have been running six days without this problem. - emckinnon
  • MINOR: It seems the phone no longer lights up when receiving a call. Seems a little strange, and this IS a beta firmware. If this is going to affect you, do not upgrade. I'm looking for an option to turn this back on - DigiGuru
    • NOTE: (Jan27/06) Grandstream has acknowledged that the fact that the phone waits until the second or third ring to light up is a timing bug and has assured me that this will be corrected prior to the production release. - thetatag
  • MINOR: The display shows AM/PM-indicator even when clock display is set to 24h-style.
  • MINOR: When adding a phonebook entry, the backlight sometimes switches off and gets switched on with the next typing of a letter, timeout-counter doesn't get a refresh by typing? - PaulK
  • MINOR: There's still a spelling error somewhere: "not alloweded" - PaulK
  • MINOR: If you hang up on a call while it is muted, the next call (either incoming or outdoing) on that line will also be muted. The flag to maintain the mute status for each line needs to reset when the call on that line terminates. - thetatag
  • MINOR: When a dial plan prefix is configured for an account it is not used when dialing from the call history directory. - Nezer
  • TWEAK: It should be enough to see into the "Missed Calls" list to let the "1 Missed Calls Press /\ to ..." frontal remindermessage disappear. - PaulK
    • NOTE:(Jan30/06) Press the center button one more time when drilling down. Although, I agree that bringing up the list should clear the message. Nezer
  • TWEAK: The "Missed Calls" list should try to resolve the numbers to names.- PaulK
    • NOTE:(Jan30/06) Press the center button and the CLID information will be in that screen. Nezer
    • NOTE:(Jan31/06) Imho, that's too late: The list should already resolve it. - PaulK
  • TWEAK: edit number allows for letters too, so if you have a number with two "1" in a row it will print a space instead of "11" .. it should allow numbers only. - PaulK
    • NOTE:(Jan30/06) What about a number that's a direct dial IP call (some_sip_address@voipprovider.net)? Nezer
    • NOTE:(Jan31/06) Hm, a way to distinguish between them would be nice. - PaulK
  • TWEAK: it's not possible to delete letters and numbers from phonebook entries, number, name etc... a delete key is missing! - PaulK
    • NOTE:(Jan30/06) Yes it is although it's not intuituve. Use the right/left cursor keys and go to the character immediatly before the one you want to delete and then press the 'mute/del' key. Nezer
  • TWEAK: an "Exit" menupoint is missing in the "Missed Calls" details of a number to get out of the GUI. Instead you have to lift the phone and down to leave the menu quickly. - PaulK
    • NOTE:(Jan30/06) Again, not really intuitive but keep pressing the left arrow key and you will return to the main display. Nezer



Firmware Notes (GXP-2000 firmware v1.0.1.13)


  • Handset volume issue - FIXED (Thankyou GS!)
  • Real Intercom, yeah baby no more workaround :) Use SIPAddHeader(Call-Info: answer-after=0)
  • Ring volume adjustment issue (adjusting ring tone using up/down doesnt appear to work (it still works if you go to the menu item it just doesnt work on the non-menu screen)).
  • New Codecs
  • some things (speed dial etc) rearranged on web interface
  • Asterisk BLF support for speed-dial buttons - oh yeah!
  • Option to hang up on disconnect (for those who were annoyed with the busy signal previously generated)
  • No more raspy DTMF when dialing
  • Now you can dial rapidly without the phone missing digits
  • Less annoying call waiting beeps

And now for the flip-side of things:

  • MAJOR: Heavy, loud, ugly echo on handset - intermittant
    • NOTE: (Nov23/05) Additional research and testing suggests that back in 1.0.1.9 and possibly 1.0.1.12 this problem was directly related to the handset volume, and higher volumes created the dreadful echo. This may be the case in 1.0.1.13 as well, although users have reported that 1.0.1.13 provided worse handset results than earlier versions. Please fix this, Grandstream! - thetatag

    • NOTE: (Nov25/05) I can confirm that this is a problem, it appears to be caused by the the volume being increased to the point where the handset mic receives what is being transmitted from the handset speaker , worst case scenario happens when two people with volume at maximum talk to each other... within a few seconds both phones will be screaming with feedback it has been a problem in every version of firmware grandstream have produced but due to a volume bug in 1.0.1.12 where maximum volume was equivilent to a setting of about 50% (unless you followed the workaround) it was more difficult to experience problems with feedback... im not sure about the best solution to this problem but it may be possible to use the "Adaptive Digital" echo canceller to solve it (without reducing the maximum receive volume) the other optioin may be to keep the microphone sensitivity setting slightly lower when on maximum volume... - SoloFlyer

    • NOTE: (Nov29/05) Further extensive testing shows cases where this problems is much worse on 1.0.1.13 than 1.0.1.9. (1.0.1.12 is just too buggy to conduct live testing.) In some cases, even decreasing the handset volume all the way did not remove or even quiet the echo. - thetatag

    • NOTE: (Dec16/05) Additional testing has suggested a work-around that is working well on one particular system and may work well for others as well while Grandstream corrects the problem in their phones. See Grandstream GXP-2000 - Solving Echo Problems for details. - thetatag

  • MAJOR: Layer 3 QoS in the "ADVANCED SETTINGS" menu has no effect. Confirmed with tcpdump. tos is always 0x48 regardless of the setting in the menu. - bani

  • MINOR: BLF remains on if a call originated from an extension where a distinctive ringtone is set.
  • MINOR: Speed dial buttons in BLF mode are disabled when the extension is in use or the BLF subscription failed! (Why?!?)
  • MINOR: After pressing TRNF, the audio cuts out and you get a nice dialtone. But after you dial the first digit, the audio stream still kicks back in until you complete the transfer.
  • MINOR: Speed dial buttons still do not work for blind transfer or 3-way conference. Properly implemented, they should do just that: speed dial — as in, behave as though you quickly dialed those digits - even if it isn't a complete number. Come on, Grandstream. Really.
  • MINOR: The HTTP configuration lets you set the No Key Entry Timeout. However, when using Early Dial, this timer is still not used. Instead an approx. 3-second timer is used and cannot be configured.
  • MINOR: Same as the last bug report, regarding the No Key Entery Timeout. This also affects transfers with the TRNF button. And because of the bug about audio reconnecting after the first digit, there is no way to know when you're still transferring the call and when it has timed out and reconnected you to the other party who is now getting all of your DTMF in their ear. Some prompt regarding transfers should remain on the screen.
  • MINOR: Phone is still prone to lockup when rebooting after firmware upgrade.
  • MINOR: Ring Volume adjustment can still only be changed though menu (it used to be the default if you just pressed up and down... please bring it back)
  • MINOR: BLF shows extension in use when the state is Unavailable.
  • MINOR: While BLF is mostly functional, it is listed as Asterisk BLF, so one would think that it should not generate warnings on the Asterisk system. Under some valid conditions and proper configuration, Asterisk reports Incoming call: Got SIP response 415 "Unacceptable Content-Type" back from xxx.xxx.xxx.xxx when BLF-related events occur.
    • NOTE: (Nov29/05) When receiving the "''Incoming call: Got SIP response 415 "Unacceptable Content-Type" error, the BLF doesn't seem to work - looks like the phones are refusing the BLF messages all together. - conexim

  • MINOR: Under certain rare circumstances, BLF indications will be left "stuck" in an incorrect state. To counter this, the GXP-2000 may need to periodically drop its BLF subscriptions and renew them. Kind of a hack for what may be an Asterisk bug, but right now the only solution is to reset the phones.
  • MINOR: Rare, but spontaneous lockups when phone is idle. (Not PoE related.) Maybe Grandstream should consider a watchdog timer that can restart modules without a full reset if something fails or hangs.
    • NOTE: (Nov29/05) Calls intermittently freeze (audio completely muted), line light stays on (blocked), and cannot be stopped by pressing any button - the call eventually times out in asterisk. - conexim
  • MINOR: (Dec12/05) When VLAN tagging is enabled everything TCP related is now broken. You must do a factory reset to get access to the web interface or get the phone to accept anymore updates. (I dont have one of these phones but i've seen this happen on lots of other hardware - can you reduce the MTU size on the phone, if so reduce it a little and see if that helps, the extra bytes from the vlan tag have a nasty habit of messing some stuff up)
  • MINOR: Placing the phone on MUTE while on a conference call not only prevents the other parties from hearing you, but also from hearing each other. You can still hear everything, including their confusion. (This may only mute one of the remote parties in addition to yourself. Additional testing may be required.)
  • TWEAK: You still cannot blind transfer a held call. You either have to pick up the line first, or do an attended transfer. (Easy to work around, but blind transferring a held call would be really useful.)
  • TWEAK: A common question in an office deployment of GXP-2000's is "What does xxx.xxx.xxx.xxx mean?" Having the IP address always displayed is nice in some environments, but it should be optional.
  • TWEAK: A configuration option should exist to turn sidetone support on or off, and possibly to adjust its relative volume.
  • TWEAK: Pressing Send to complete transferring a call should be optional, and coupled with the Early Dial configuration option. If the technology does not allow this, at least the timeout should be used to determine when you are done dialing.
  • TWEAK: There is currently no way to see if the phone is forwarded when call forwarding is enabled locally.
  • TWEAK: When in a conference call, pressing one of the conferenced lines should immediately put the other on hold and connect just the pressed line. You should not have to quickly press hold and then the line you want to talk to.
  • TWEAK: When the phone is in Auto Answer or Auto Answer via Call Info with rings set to 0, the phone still produces a half-ring before picking up. Notification of the incoming page should be configurable or non-existant (leaving it up to a PBX). Configurable would be a better option as a PBX will not always be in the middle of some connections.



Firmware Notes (GXP-2000 firmware v1.0.1.12)


  • The automatic reboot after firmware upgrade appears to cause the phone to lockup... all lights inc backlight remain on and nothing appears on screen... hard restart ( disconnect reconnect power ) solves the problem and everything works perfect from then on - very annoying if upgrading multiple phones... Upgrading via HTTP didn't seem to cause this issue for at least one user.
  • Ring volume adjustment issue (adjusting ring tone using up/down doesnt appear to work (it still works if you go to the menu item it just doesnt work on the non-menu screen)).
  • Handset volume issue ( Sometimes the volume setting at full becomes equal to that at 3 or 4, making the handset very quiet. Doing an Audio LB test or switching to and from speakerphone seems to fix this problem temporarily)
  • Speakerphone is now usable with AEC. WOOHOO! :)
  • Still no intercom but the workaround works fine so who cares...



Firmware Notes (GXP-2000 firmware v1.0.1.9)


  • Speakerphone Suffers from signifigant Acoustic Echo
  • Still no real intercom system... I have been able to work around this by setting up a seperate user account on line4 which has auto answer enabled :)





Created by: thetatag, Last modification: Tue 11 of Sep, 2012 (00:39 UTC) by admin
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