Page Contents
- Firmware Notes ("Beta" 1.1.1.13):
- Firmware Notes ("Beta" 1.1.1.12):
- Firmware Notes ("Beta" 1.1.1.11):
- Firmware Notes ("Beta" 1.1.1.10):
- Firmware Notes ("Beta" 1.1.1.9):
- Firmware Notes (Beta 1.1.1.7):
- Firmware Notes (Stable 1.1.0.16):
- Firmware Notes (Beta 1.1.0.13):
- Firmware Notes (Beta 1.1.0.11):
- Firmware Notes (Alpha 1.1.0.4):
- Firmware Notes (Alpha 1.1.0.1):
- Firmware Notes (Beta 1.0.2.13):
- Firmware Notes (Beta 1.0.2.8):
- Firmware Notes (GXP-2000 firmware v1.0.2.6)
- Bugs / Concerns:
- Firmware Notes (GXP-2000 firmware v1.0.2.3)
- Firmware Notes (GXP-2000 firmware v1.0.1.13)
- Firmware Notes (GXP-2000 firmware v1.0.1.12)
- Firmware Notes (GXP-2000 firmware v1.0.1.9)
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 ("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?
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
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.
(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
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
- 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
- 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
- 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
- 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
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 | ~netmask)) 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)
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
Build 1.1.0.12 5/15/2006
Changelog listed above taken from Grandstream 1.1.0.13 Release Notes PDF included with 1.1.0.13 firmware -benny23
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:
Changelog
Build 1.1.0.11 4/28/2006
Changelog listed above taken from Grandstream 1.1.0.11 Release Notes PDF included with 1.1.0.11 firmware - flu
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
- This is difficult to verify. Basically our 3-way conferencing can be broken into the following steps:
- 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'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 "NOTICE6998 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
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
- 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
- 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 t