Firmware Summary

Firmware Notes 1.1.4.18:

• 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
• 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

• 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

• 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

Firmware Notes 1.1.4.17:

• 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:

• 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:

• 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" 1.1.3.2):

Bugs / Tweaks (1.1.3.2):

• Major : (Mar20/07) Audio Quality is terrible on PCMU (HW version 0.4). Volume is extreamly loud on a standard PRI tuned to a 102 mW Test Line. I have to turn the volume all the way down to get anything remotely acceptable. Various clicks, beeps on every call. - ninthclowd
  • Note : (Mar24/07) (HV=0.4) Volume is loud and very clear and I like it. l have several older users who appreciate the loudness!. - anthony
• Major : (Mar22/07) Audio Quality is terrible on G729 (HW version 1.1). Volume is extreamly loud on a standard PRI tuned to a 102 mW Test Line. I have to turn the volume all the way down to get anything remotely acceptable. Various clicks, beeps on every call. - jozwikjp
• Major : (Mar22/07) (HW version 1.1) Phone constantly locking up in the middle of a call or in the middle of anything or any other situation. using AC adapter. unplug/plug required to reboot. - jozwikjp
  • Note : I have this happen quite a bit on certain phones using PoE on certain cable types (cat 5 instead of 5e or 6). It might be worth it if you try changing out the cable or having the line tested for signal loss. Easiest way to see if that's your problem is by turning off PoE on the switch port going to that phone and using the AC adaptor. -ninthclowd
  • note : (Mar22/07) I'm also getting lockups with this firmware, using the AC adapter. Mine is HW version 2.0 as well. The lockups I have had have been when the phone is idle, but it very well might lock up on calls as well (I haven't used it enough to know). -csnyder
• Major : (Mar27/07) Call drop when make supervised transfer and quickly select hold line: We have 3 concurrent call (eg line 1 speaking and 2,3 hold) when perform attendant transfer from line 1 to line 2, and quickly select line 3 the phone "freeze" line 3.

• Major : (Mar27/07) Handset still hisses and the screen still blanks. However the artifacts and randomness of the display haven't shown up yet. - vgster
  • Note : Agree, can hear hisses nearly every second making the conversation sound terrible and have repeatedly interrupted phone calls with this update.Please FIX this Grandstream ! Phone cant be used like that in a professional environment -Datu
  • Note :I have noticed screen blanking, when using PoE. Phone doesn't even wake. That's with nortel switch, on Cisco it worked, but it vas HW version 1, this one is 2. Maybe that is a power trouble? -bad2Dbone
• Major : (Mar27/07) Screen blanking as bad as before (making phone almost unusable), HW0.3, haven't noticed any problems with sound quality as others have reported - CG
• Major : (Mar27/07)Very disappointed. Agree with above. screen blanking and audio problems. can we have a reason why these problems still exist? Every firmware release with these faults damages our business and GS sales. When will they be fixed? - Richard
  • Note : Agree, my boss is drivng crazy with these never ending problems with this phone. -Datu
• Major : (Apr06/07) Rebooting does not work using Webinterface, Phone remains in undefined state, Menu is still accessable but reboot can only be done by interrupting power cycle, message on screen shows 'preparing reboot...' permanently till hard reset. MAC000B82083067 - Datu
• MINOR : (Apr12/07) Disabling Silence supression is not working on older hardware versions. Minor but very very anoying, older hardware will not disable silence suppression, you never know if the remote party is still there. - Joao Carvalho

Firmware Notes ("Stable?" 1.1.2.27 01/30/2007 - Not yet available for public use):

• Note (May08/07): Their support sent me this firmware via mail on 08/may/07 saying that it is the latest official firmware, but this firmware cannot be found on their firmware download page! - QSS (it)

Firmware Notes ("Alpha/Beta" 1.1.2.26 - Not yet available for public use):

• * Same problems for us: Could we have some fixes please Grandstream?
• We really are desperate - will need to replace 30+ units unless handset hiss is overcome. Any estimation of time will help.
• Note (Mar19/07): Should be able to downgrade to the 'stable' 1.1.1.14, despite what some of the notes say. This does not have the hiss on my units.

Firmware Notes ("Beta" 1.1.2.25):

Bugs / Tweaks (1.1.2.25):

• MAJOR: (Dec20/06) REGISTER EXPIRATION After boot and successful registration I loose the registration after the Register Expiration time (default 60 minutes). Workaround: set Register Expiration to 65535 minutes. - kam
  • Note : (Dec29/06) I can confirm this, although it does not happen with every provider. For me, it happens with dus.net, always after about 1 hour - other providers are working fine...Update: It still doesn't work for me with v1.1.2.25 - Mirak
  • Note : (Dec30/06) For me, it happens with xs4all.nl and only since firmware version 1.1.2.23. SYSLOG shows "SIP/2.0 481 Dialog does not exist ..." and "XS4ALL: REGISTER rejected by 481, retry in 15seconds" - kam
  • Update: (Jan19/07) Same reregister problem still happens with v1.1.2.25 depending on provider. - kam
  • Note: (Jan23/07) Using "sip debug" on asterisk, I've found that the only difference is when 1.1.2.x tries to renew registration, it sends the REGISTER packet with authentication attached. By contrast, earlier versions send REGISTER without authentication, which provokes a "401 unauthorized" response, then the phone sends another REGISTER with authentication, and that second REGISTER succeeds. In the first case (1.1.2.x) the first (renewing) REGISTER packet has authentication, and it gets "481 Call/Transaction Does Not Exist" IF Asterisk is configured with pedantic=yes in sip.conf. If asterisk is configured with pedantic=no, the REGISTER messages always succeed. Clear as mud? Perhaps someone who knows more about SIP might be able to explain the difference. - bcheath
    • Update: (Jan26/07) OK, I was wrong about the registration. The actual difference is that 1.1.2.x includes a "tag" on the To: header of the REGISTER packet when it goes to renew registration. Asterisk (with pedantic=yes) considers that invalid. - bcheath
      • Update: (Jan26/07) I sent this information to Grandstream support, and they replied that they found the problem and the fix will be in the next release. - bcheath

• MAJOR: (Jan26/07) Screen blanking With HW0.3, screen blanking is worse than ever. I can be almost sure that the screen is blank if I've been out for an hour and there has been 1-2 phone calls. No key seems to get it back again (mute/del and sometimes some other keys used to get it back) so reboot necessary and a lost list of calls. This is getting silly, please solve this problem now Grandstream! carl-g
  • Note : (Jan29/07) Agreed. GS broke the screen sometime after 1.0.1.13. It worked perfectly before that so we know it can work right. I really think we deserve an explanation (a good one) from GS on this matter now. It's not even right on 1.1.1.14 'stable' release. I wouldn't mind so much if 1.0.1.13 had all the core features working and you could go back to it - jedi98
  • Note : (Feb08/07) Yes, The phones are painful with this fault. It did definitely not occur when the units were purchased, otherwise they would have been returned to the supplier. Please fix. rdman

• MINOR : (Dec20/06) Switch/Router Since the upgrade, I can no longer setup the integrated switch/router. The neccessary options are just missing. If I look at the HTML-Source, all of them are still there, but commented out... Can anyone confirm this? Update: v1.1.2.25 is still missing the router functionality - Mirak
  • Note : (Dec20/06) Confirmed. I successfully saved the config2.htm file, uncommented the router code block, changed form action to the phone's IP (http://192.168.1.X/update.htm), and submitted the changes. They were reflected in the updated commented code. I'm guessing the functionality is still there, just an oops on the released code. - Shane Steinbeck

• MINOR : (Jan16/07) Handset hiss It's still there. They say they have fixed the hiss from the other parties handset but how about the hiss when you aren't conencted to any other parties? The handset hisses. - vgster
  • Note: (Jan22/07) I noticed that when the handset volume is at the lowest level almost no hiss is heard, but one volume level higher the hiss starts. - pnxs
  • Note: (Jan25/07) I also noticed this - my phone is now constantly on the lowest setting but it suits me fine! Overall audio is far better in this firmware vs. the old firmware I had. - drsox
  • Note: (Jan31/07) I get the hissing, as well as a regular clicking sound (approx 1 per sec), reducing the handset volume doesn't remove the clicking. - emdee
  • Note: (Feb05/07) I hear hissing, clicking (approx 1 per sec), along with various intermittent, low volume, high-frequency BEEPS of various pitches. It's really odd. - engineer_dan

• MINOR : (Jan16/07) GSM_CODEC No problems with Asterisk 1.0 , but Asterisk 1.2 and 1.4 have problems with musiconhold using grandstream's GSM codec. Sound is choppy. I thought it was Asterisk problem, but when i tried to connect from a GXV-3000 to a GXP2000 it did't work with the gsm codec. I don't know if it is Grandstream's fault or Asterisk's, but all other phones (Cisco , Polycom, sipura , etc ..) work OK - Joao Carvalho

• MINOR : (Jan18/07) NTP Server Using a fixed ip address seems do not work. GXP never send any UDP port 123 packet to our NTP server (Linux openntpd), i checked with iptables -j LOG. It worked in previous beta, did they broken something? - agx
  • Update: (Jan22/07) Confirmed numerical IP address entry of the NTP server does not work, need to enter the DNS name. - emdee

• MINOR : (Jan22/07) Can't dial using # With this version, any dialed number containing # doesn't work. The phone says "404 NOT FOUND" and plays a busy tone. Previous versions (1.1.2.23 and earlier) do not have this problem. I use # in my dial plans so this is a problem. - bcheath

• MINOR : (Jan23/07) Can't dial using # With this version, any dialed number containing # doesn't work. The phone says "404 NOT FOUND". This seems to be because the phone sends %23 instead of # so the NOT FOUND is because Asterisk etc does not find the ...%23... in the dialplan. Workaround: I had an extension ## in the dialplan, so in extensions.conf I had entries like: _##... (must have pattern if using #). I added an identical extension but changed it to: %23%23... and that works even if it is an ugly Grandstream-patch. carl-g
  • Note: (Jan23/07) Yep, I can confirm that this is what's happening. So it seems to be Asterisk's fault in this case. The INVITE message is a URI, so the % coding is valid, I think. - bcheath
    • Update: (Jan23/07) Solved - I put "pedantic = yes" in sip.conf in the [general] section, and Asterisk now properly decodes the URIs (I have 1.2.14) - bcheath
      • Update: (Jan26/07) Thanks, works well. I didn't realize that the %23 was a valid way of doing it so never looked at my Asterisk. But you're right, it works well with pedantic=yes. carl-g
        • Update: (Jan26/07) With the caveat that pedantic=yes causes the phone to lose registration, due to a problem with the way that 1.1.2.x re-registers (see above). - bcheath

• MINOR : (Jan29/07) In call DTMF key display not turning off: Disable in-call DTMF display has been set to yes but inspite of it, when digits are pressed, whole screen clears and single digit is shown and sometimes if numbers are entered quickly two three numbers are shown but only momentarily since they go away quickly. Would be nice if numbers don't show up at all. denpun

• MINOR : (Feb08/07) Phone fails to operate when it does not recieve any DNS server addresses from the DHCP server: I don't have any DNS servers on my network so I had to put some imaginary ones into my DHCP config to make my phones work. This was never a problem with software versions <= 1.1.1.14. - dpnss_user

• MAJOR : (Feb19/07) Phones internal web server appears to serve partial pages ... page ends with uncompleted HTML tags, refreshing causes breaks at different places on the page. Has anyone else seen this? - emdan
  • Note: (Feb21/07) This problem has been around on and off since the original FW releases. Basically it's really fussy about packet loss, timing, VPNs and MTU. I believe that the original fix was to do with MTU. Usually can be worked around by page reload or re-login. - jedi98

• MINOR : (Mar06/07) Phone drops packets when accessed via wireless LAN. Pinging other devices on the network I have no lost packets, but the GXP-2000 is dropping 90%. The web page is inaccessible. It works fine from wired LAN. This worked fine immediately before upgrade, and broke immediately after. - naftali5

Firmware Notes ("Beta" 1.1.2.23):

Bugs / Tweaks (1.1.2.23):

• MAJOR: (Dec28/06) IDLE_SCREEN Setting HTTP to download gs_screen does not work. There are no calls for gs_screen.xml to the httpd daemon on my linux box. I do see calls for gs_phonebook.xml - Anthony

• MAJOR: (Dec27/06) VLAN Does not work in this version. If VLAN ID is defined will not gain network, a reset to defaults has to be performed to regain network functionality. Downgrade to 1.1.1.14 for VLAN functionality. - Joao Carvalho
  • Note : I confirm this, ethereal analyze has shown absolutly no network traffic (except ethernet physical link negociation certainely through the GXP2000 ethernet chip) after reboot. If config menu is locked, then it is not possible to make a factory reset. Your GXP2000 cannot be unlocked and need factory return to reprogram the Flash chip. - Olivier.

• MAJOR: (Dec20/06) SCREEN BLANKING Guess what? No prizes for guessing... the screen still goes blank, HW:0.3, when you turn your back for a couple of minutes. Never does it when you're watching, it's far too sneaky for that!! Blanking is a lot more frequent than previous FW, approx 10 min. - Jedi98 Worse than ever with this version! - Andrew

• MAJOR: (Dec20/06) REGISTER EXPIRATION After boot and successful registration I loose the registration after the Register Expiration time (default 60 minutes). Workaround: set Register Expiration to 65535 minutes. - kam
  • Note : (Dec29/06) I can confirm this, although it does not happen with every provider. For me, it happens with dus.net, always after about 1 hour - other providers are working fine...- Mirak
  • Note : (Dec30/06) For me, it happens with xs4all.nl and only since firmware version 1.1.2.23. SYSLOG shows "SIP/2.0 481 Dialog does not exist ..." and "XS4ALL: REGISTER rejected by 481, retry in 15seconds" - kam

• MAJOR: (Dec21/06) TRANSFER PROBLEM If attended transfer fails the line that was supposed to be tranfered goes on hold (led blinking) and call cannot be resumed, screen looks like the phone is on-hook (time and date displayed). When the other party disconnects, led stops blinking and line can be used again to make a call. To recreate the problem: Receive or call on LINE1, press LINE2 to make another call to ext that can't be xfered eg. echo, press transfer and press LINE1 to complete. TRANSFER FAILED is displayed and see for yourself. - Pietia

• MAJOR: (Dec26/06) Router Before upgrade, I used the pppOE function combined with the router function. My router had the IP 192.168.1.1 After the update, the GPX cannot any more accessed over the web interface. If you change the IP as a static one, it comes not up the network. Only a factory reboot helps. I figured out, if the switch mode is on, no problems with the firmeware upgrade, only it the router mode is on. - edonia

• MINOR: (Dec20/06) NTP Time Sync has not worked since the upgrade to this FW, with both windows and linux time servers. Both servers working for all other clients. - Jedi98
  • Note : (Dec20/06) NTP Time Sync Seems to be fine for me. I'm running NTP on my Asterisk box on the same subnet as the phone. IP address is used for the NTP server address on the GXP.- Shane Steinbeck
  • Note : (Dec20/06) In another forum (ip-phone-forum.de, german) there are also some reports about ntp no longer working. For some people using another server or entering IP or hostname helped. - Mirak
  • Note : (Dec20/06) Specifying a hostname instead of an IP address in this field solved the problem for me. Previous versions worked with both. - job
  • Note : (Dec20/06) Confirmed, using FQDN fixes this. I guess gethostbyname() is broken. - Jedi98
  • Note : (Dec21/06) For me NTP works (with IP address or FQDN) only if i put a valid DNS server in basic settings page. I tested this with BT200 and GXP2000 phones.- Francesco_r
  • Note : (Jan09/07) It only works for me via FQDN, previous versions worked fine by IP. - rdman

• MINOR : (Dec20/06) Switch/Router Since the upgrade, I can no longer setup the integrated switch/router. The neccessary options are just missing. If I look at the HTML-Source, all of them are still there, but commented out... Can anyone confirm this? - Mirak
  • Note : (Dec20/06) Confirmed. I successfully saved the config2.htm file, uncommented the router code block, changed form action to the phone's IP (http://192.168.1.X/update.htm), and submitted the changes. They were reflected in the updated commented code. I'm guessing the functionality is still there, just an oops on the released code. - Shane Steinbeck

• Note : (Dec20/06) Audio Quality Vast improvement in audio quality and clarity on, incoming on g.729 and g.711 compared to all recent versions. - Jedi98
  • Note : (Dec20/06) Audio Quality Agreed, although there is an annoying hiss/noise on the handset, it's much easier to deal with that the constant echo I had.- Shane Steinbeck
  • Note : (Dec20/06) Audio Quality Confirmed: I also have annoying hiss, maybe this is 'comfort' noise. - Jedi98
  • Note : (Dec21/06) Audio Quality Confirmed: VERY annoying hiss, compared to 1.1.1.14 it's terrible. There's nothing 'comfort' about it. - Pietia
  • Note : (Dec21/06) Audio Quality Yes, there is a noise in the handset but it's not so terrible, i prefer this that echo. The handset volume is also nicely loudest than previous versions, perhaps too - Francesco_r
  • Note : (Dec22/06) Audio Quality Confirmed: I rolled back to .14 because of it. But then again I tweaked my Asterisk box to compensate for the echo so echo isn't as much of a deal to me compared to the hiss - ninthclowd

• ATTENTION : (Dec20/06) After the update. the phone could no longer connect to the network. I had to do a factory reset to be able to access it again... There are also reports from other people that had the same problem. - Mirak
  • NOTE: Grandstream has stated that the router function of the GXP-2000 has been disabled in this Beta version. An improved router function is in the works and will be enabled again in a forthcoming version. If you absolutely must have the router function, do not participate in this particular Beta! - thetatag
  • NOTE: This has nothing to do with whether or not you use the router function of the GXP-2000. I have experienced the same thing. I found that it had to do with the configuration files I was sending the phone. They were the 1.1.1.7 config files. I haven't figured out what needs to be commented out in order for my TFTP config files to work with this phone yet, but I will post if I find out. - ninthclowd
  • NOTE: I found out today from Grandstream that this was happening to me because I had set my Layer 2 QoS. 802.1p priority value to something other than zero. I don't know if this is why it happened to everyone else but I figured I'd post this. - ninthclowd
  • NOTE: It is possible this is related to the VLAN problem i have reported before. Under no circumstances set the VLAN ID. Keep it 0. - Joao Carvalho

• MAJOR: (Dec20/06) BLF Lights Same issue as before, when paging to 50 phones the side car doesn't report the correct status for the extensions. It flashes for a few, stays red for others etc... it just goes crazy, this doesn't seem to happen on small paging groups e.g. 5 extensions. I was really hoping this would be fixed by now. - JoeKane
  • Note : (Dec20/06) BLF Lights It's highly unlikely that this will ever be solved, for two reasons. 1) Paging that many extensions is insane. Many customers we have that try to page more than about 30 phones tend to bring their network to it's knees by doing so. 2) I doubt that the GXP platform has the horsepower to actually receive and process that many NOTIFY packets. If you want that many BLF lights to work during paging, try ShoreTel, but don't expect anything less than a gig network to handle the load. - Galen
  • Note : (Dec21/06) BLF Lights I do about 10 pages a day to 50 phones without issue. If the phone cant handle the load then can Grandstream comment on the maximum amount of paged BLFs the side car can support???. - JoeKane
  • Note : (Dec21/06) BLF Lights I reduced the page to 30 phones and have the same issue, I will try 20 later. - JoeKane
  • Note : (Jan03/07) BLF Lights I reduced the page to 10 phones and have the same issue, Seems to work with about 5 / 6 extensions, If the phone is unable to handle the load is there anyway the phone could be refreshed without having to reboot everytime? - JoeKane
  • Note : (Jan04/07) BLF Lights Really the issue we've noticed --at least with Asterisk-- is that it's not necessarily an issue with the GXP-2000 or the sidecar, it's an issue with the network itself. I would suggest that if the BLF lights "stick" it could be either a buffer/not enough memory allocated in the phone for handling SUBSCRIBE messages sent to the phone, or that the hardware platform is simply not robust enough. But more than likely, it's actually with the network that the phones are plugged into. One of our customers bought a whole mess of GXP-2000s from us, and they worked just fine, except when paging 30 or more phones. Their 100Mbit network just didn't seem able to deliver packets quickly enough, so Asterisk assumed a timeout, sent more packets, and within about 5 seconds, Asterisk was flooding the network with repeat packets because it wasn't getting responses from the phones quickly enough. - Matt Blecha
  • Note : (Jan04/07) BLF Lights Thats interesting alright so you have a setup where the sidecar can report a 30 BLF phone page correctly? I have 3 Dell POE switches all feed from 1gig connections and 62 phones. If it is a network / asterisk issue then surely the Snom 360 sidecar would experience the same issue???? which in my case it doesn't its been in production for 6 months without issue..... - JoeKane
  • Note : (Feb01/07) BLF Lights Sorry for the long delay, we've been quite busy with our Asterisk CC apps. We haven't seen the issue much with other manufacturers, it seems that Asterisk will broadcast the packets, and the Grandstream doesn't respond fast enough, so Asterisk broadcasts another mess of packets to the phones that haven't responded, and the Grandstreams ignore them again, causing a loop of sorts. You also need to remember that we're not just talking about SIP signalling here, but the RTP, too. Each SIP/RTP channel running is about 64k each. (Assuming G.711) That might not seem like a whole lot, but it can add up quick, when Asterisk is retrying to establish those channels missed. - Matt Blecha

Firmware Notes ("Stable" 1.1.1.14):

Bugs / Tweaks (1.1.1.14):

• MAJOR: (Dec15/06) No reconnect during ip renew from dynamic dns after 24h timeout: Since I updated the firmware to 1.1.1.14 the phone doesn't reconnect after my ISP changes my IP. Any ideas? - codeworker

• MINOR: (Nov30/06) Have to Hit Line Button Twice: When you hit the transfer button it tells you to do a blind transfer or hit a second line button to do an accounced transfer. You have to hit the second line button twice before it switches to that line. -Joshua

• MAJOR: (Nov29/07) No handset audio when off hook: On occasion - perhaps 1 out of 10 times the handset is lifted - the speaker in the handset is not enabled. The phone says it is off hook, and calls can be placed, but there is no audio. This is with the "original" hardware revision only. Phones in my office with the more recent hardware (with dual color LEDs) are not having this problem.

• MAJOR: (Nov30/06) No speaker audio (ringtone) when headset connected: When you plug in a headset, the audio output is redirected to the headset. Even the ringtone, which should be optional imho. I would like to have the option in the config and by default the ringtone on headset AND the speaker. - codeworker
  • Note : (Dec01/06) This seems to be an hardware issue. I mean that AFAIK there is no way to direct ringing tone to the speaker while headset is connected, because the connector has a switch inside it that cannot be overridden by software. This topic has already been discussed some months ago, search further down the page. I don't know if newer HW revision have software-selectable audio path to override this limitation. - Kurgan

• MAJOR: (Nov16/06) 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. - stegie
  • WORKAROUND: Go on other lines, then come back on your onhold one and they will be able to hear you again. - stegie

• MAJOR(Nov10/06) 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 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. The bug is STILL present in 1.1.1.14. Jaroslav Janacek
  • NOTE : (Dec 01/06) Not sure but I think I have encountered this. The phone wouldn't work at all on a particular IP. I changed the IP to something else and it was fine. - Chewi

• MAJOR: (Oct16/06) MTU Discovery The web interface for the GXP2000 has the DF bit set in outgoing packets presumably so that it can do MTU discovery but when it gets an icmp reply asking it to reduce packet size, it just stops sending packets. I have only tried with mtu set to 576. Tcpdump Log - SoloFlyer

• MAJOR: (Oct20/06) SCREEN BLANKING old man to young boy: Son, i remember a time when GXP2000's didnt randomly go blank... Really grandfather that must have been a wonderful time! - SoloFlyer

• MAJOR: (Nov01/06) No RTP Keep-Alive during Mute I'm not seeing RTP keep-alive packets after the Mute button is pressed, in the latest firmware 1.1.1.14. I would expect an RTP keepalive packet to be sent while Muted at the same interval as the SIP keepalive. - awint
  • Note : (Nov02/06) . We discovered this problem over a week ago and I've been trying to find a fix. This was listed as "fixed" in FW v1.0.2.13 but I saw it in 1.1.0.14. Just tried 1.1.1.14 before reading this post and see the problem is still there. - et

• MAJOR: (Oct26/06) Handset echo Using ulaw in GXP-2000 -> Asterisk -> PSTN or another * server gives major echo on the handset. It's not like PSTN echo, it's much more distinct with a longer delay. I'm wondering if it's only in the newer "heavy" handsets or on all of them. I have at least 2 phones displaying this problem and it is 1.1.1.14 related. Is this the same as some of the other audio quality issues discussed in the notes? Anybody else hearing this? Anybody else hearing this? -Shane Steinbeck
  • CONFIRMED : (Oct26/06) I have the same issue, very clear and delayed echo of my own voice using Asterisk 2.0.7 Bristuffed and one ISDN BRI with zaphfc. I have no echo if I use voip instead of ISDN. Have you tried using a different codec? - Kurgan
  • NOTE : (Nov 12/06) I have PSTN on X100P/X101P cards - echo was annoying. I cranked tx up to 10 in /etc/asterisk/zapata.conf and now I notice an echo only at the beginning of conversations for a few seconds. For further info, try http://asterisktutorials.com/videos/echo/movie.html - Anthony
  • NOTE : (Nov 14/06) Nice tutorial. I have discovered that I had maybe a too high RX (hits full scale all the time), so I decreased it a little, but I cannot increase TX, no matter what value I write to "txgain", my TX level is below mid-scale. I am using an HFC-based ISDN card. - Kurgan
  • NOTE : (Nov 15/06) Yes, I'm getting this echo also, only on the phones with 1.1.1.14. Other phones (GXP-2000s) are fine and have no echo with the older firmware. Really hating these phones. - njtaz76
  • NOTE : (Nov 15/06) I find it very interesting that you say you have no echo problems on the old firmware. I had assumed that the echo issue was unavoidable. All my phones are on 1.1.1.14. I would love to get rid of this problem. Anyone for upping this to MAJOR? - Anthony
  • NOTE : (Nov 15/06) I'll pull the trigger and call it major. I get it on VOIP to VOIP with various codecs, echo shouldn't be an issue in pure digital environments. - Shane Steinbeck
  • NOTE : (Nov 21/06) Grandstream might think about giving us some configurable options in the sound modulation code, maybe different ways of handling sound for PSTN, vs digital, echo cancellation, gain etc. These options could be configured on the fly with SIPAddHeader. - Anthony

• MAJOR: (Feb26/07) Hook Status After a call has been transferred, lights for both lines will go out, then line1 will light up and you will get a dial tone, if you hang up at the exact time that line1 lights up the light for line1 will be lit up but the handset will be hung up, 90 seconds later the light will go out and the phone will go back to normal, But if you take the handset off hook the handset will only remain off hook 90seconds. After that amount of time the phone will release the line and will register as on hook even thought the handset is still off hook. If a call is made or received on another line while line1 is lit up, then when line 1 times out, the call you are currently on will be replaced by the engaged tone and you will not be able to hear the remote party, although the remote party can still hear you until you hangup. - SoloFlyer

• MINOR: (Nov03/06) MWI - Bug on 100+ New Messages When the MWI indicates 100 or more new messages, the %d NEW MESSAGES display (when the phone is off-hook) only shows 2 "digits", with the first being non-numerical past 99. To duplicate, set the number of New messages to 100, and take the phone off-hook. I see ":0 NEW MESSAGES" at 100 -awint

• MINOR: (Oct03/06) PROVISIONING LOOP Keeps downloading and provisioning repeatedly. It appears to do this twice, then be ok, but as soon as any changes are saved to the configuration (via the web interface) it provisions twice again before "settling down". -Mike
  • Note : (Oct09/06) I did not encounter this bug. The firmware docs inform us that the phone will download the firmware in two reboots. The upgrade worked smoothly. I can change options in the WebUI without the phone rebooting by clicking "update". - emdee

• MINOR (Oct11/06) "No Key Entry Timeout" change problem Advaced settings/No Key Entry Timeout doesnt change after update (still 4 sec) i need at least 20 sec for number dial - cervajs
  • Note : (Oct13/06) Confirmed bug. - Anthony
  • Note : (Oct14/06) Partially confirmed: Setting the value to 20 as mentioned here does not work for me too - it's silently reset to 4. However, setting it to 10 seems to work as expected. So I suppose Grandstream is just trying to only accept values within a "normal" range - Mirak
  • Note : (Oct17/06) it does not matter what value I use, it resets internally to 4 seconds - Anthony
  • Note : (Oct18/06) I did some tests and apparently the maximum value is 15sec - Schnuffle
    • Note : (Oct18/06) As far as I know, this has always been the phone's behavior. We have phones running on 1.0.1.9 that do the same thing. This sounds more like a feature request than a bug. - NateBell
  • Note : (Oct18/06) My bad. I also did the tests and 15 seconds is the maximum and that is just fine for my users - Anthony

• MINOR: (Oct18/06) BLF leds stay on It seems that sometimes BLF lights stay on after the extension that was busy has hung up, and require a reboot of the phone (the one that has the BLF light struck on, not the one that the light refers to) to make the light work again. I have only 8 phones and it happened only two times, so I am not sure about this bug, but it never happened with 1.1.0.16 fw. Does anyone else experience the same issue? - Kurgan
  • Note : (Oct27/06) It may be an Asterisk issue. Check "show hints" on Asterisk CLI to see what the "watchers" see. I have Asterisk 1.2.12.1. I am not certain yet whether such issues are Asterisk related or GXP2000 related. - Anthony
  • Note : (Oct28/06) I use Asterisk 1.0.7 (Debian) and it seems that I don't have the "show hints" command. Anyway, since rebooting the phone with the faulty BLF light solves the issue without touching Asterisk, and restarting Asterisk does NOT solve the issue, I doubt it's an Asterisk issue. - Kurgan
  • Note : (Oct29/06) If you do a "reload" on Asterisk while the lights are on, they will stay on to the phone re-registers with Asterisk. - mtryfoss
  • Note : (Nov05/06) This may be an Asterisk issue. Try to check asterisk for messages like "Incoming call: Got SIP response 415 "Unacceptable Content-Type" back from (IP)". BLF status is sometimes sent with content-type:unknown. - guri
  • Note : (Nov17/06) Im having this issue with the sidecar and on phone's, Paging really sends it into a spin, but after about 15 minutes it seems to refresh itself, is there a possible refresh command for the hints so the status can refresh???. - JoeKane
  • UPDATE: (Nov22/06) It appears the phone loses it's subscription for the hint (not seen in 'sip show subscriptions'), yet 'show hints' still shows that there are the same number of watchers. Asterisk will continue to sent NOTIFYs, but the phone will reply with "Unacceptable Content-Type" until either the phone decides to re-SUBSCRIBE or the phone is rebooted. This issue has been less frequent since 1.1.1.x firmwares. - iwes

• MINOR: (Oct23/06) AUTO-UPDATE RESTART sometimes the GXP2000s will restart for firmware updates while the phone is being used to make a phone call... as a workaround you can disable the automatic checking for updates or setting it to very high intervals -SoloFlyer
  • Note : (Oct29/06) I confirmed this is happening also. It seems to be revisioning every 5 minutes or so, even when a call is in progress. I have the time set to check every so many days, not every 5 mins. -njtaz76

• MINOR: (Oct24/06) Dialed number disappears if I receive a call while dialling If I am entering a number to dial, and another call comes in, the phone shows the informations about the incoming call, and removes from the display the number I was dialling. Even after the other call has been answered by someone else, I don't get back the dialling display. The numbers I have entered have been accepted, and I can dial the last numbers and press SEND to call, the only issue is that I don't see the numbers I have entered, but this is disturbing. I suggest that if a call comes in while I am dialling, the new call rings and the line button flashes, but the display does NOT switch to show the information about the incoming call and stays in "dialling" mode to allow me to finish dialling properly. -Kurgan

• TWEAK: (Oct13/06) SIP MESSAGE After Grandstream added SIP MESSAGE - support, it would be nice, if they added support for the Content-Dispositon = desktop Header or an option to select the output location of the text (not hidden in the User Menue). Background: This Feature would be very useful for status-information. The sent message appears in the last line of the display and can be deleted by sending an empty message. Should be not much of a problem to also include a new variable for the XML idle screen to position the text. - bladerunner

• Note : (Edited Oct19/06) Mic volume I have upgraded from stable 1.1.0.16 to this firmware. I have found that 1.1.1.14 seems to have improved the mic sensitivity volume level for the handset and speakerphone. Outgoing calls over g729 are still causing some problems (breakups, volume level) that are not seen on Snom300 phone's on the same network. I no longer think this firmware has resolved audio quality issues with this phone. - emdee

• Note : (Oct12/06) Audio Quality is markedly improved compared to 1.1.0.16, when using g.711 codec. AGC in the prior (1.1.0.16) firmware overshot wildly, making the volume of callers and locally generated music-on-hold fade in and out. I'm delighted with the improvement. - (EDIT) Like freman (below), I may have spoken too soon. Although the quality of received audio is vastly improved, issues remain with wildly varying automatic gain control action in the TRANSMITTED audio. To my ear, it sounds as though gain reduction is overshooting in the time-domain during and following an over-threshold transient. Possible solutions: Reduce the COMPRESSION RATIO and/or change AGC to a slow, window-comparator, feed-forward design (so limited processing speed will not result in constant "hunting" in the gain reduction). If successful, these changes will result in more consistent transmitted volume and thus permit an increase (3-6 dB) in transmitted audio without inviting dreaded clipping/echo problems. Getting a little louder will go a long way to improving the perception of audio quality. </soapbox mode> - engineer_dan

• Note : (Oct13/06) Audio Quality I gave up on g.729, and tried GSM, according to the people I was speaking with it waved in and out and generally sounded bad. I switched to PCMU (ulaw) and everyone says it sounds fantastic. - (EDIT) I spoke to soon, I get to speak to a lot of people (it's a helpdesk phone) and some of them are reporting that I'm breaking up. The network path is almost completely free so I dowbt it's congestion - freman

• Question:(Oct10/06) SIPS Is SIPS already implemented in this firmware version or should it be added to the Feature Request List? - kodomo

• MAJOR (Sep26/06) Paging Issue In 1.1.1.14, 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. This does not happen on my Snom 360 extension unit, can you please arrange a fix for this. (Phones on 1.1.0.16)??? - joekane

Feature Requests

• FEATURE REQUEST: (June 28/07)
• Big thanx (Jun28/07) for "speed searching phonebook".
  • Will be nice to see the name of caller on the screen instead of his number, if the caller is in the phonebook

• FEATURE REQUEST: (June 04/07) Phone book browsing enhancements
  • Add a key-repeat to the arrows so you can simply hold them down to scroll through the list.
  • Allow the keypad buttons to jump to phone book entries beginning with those letters. - RyanWilliams (au)

• FEATURE REQUEST: (May 07/07) Automatic idle screen update
  • Automatic download of Idle Screen (not by user request). It would be useful to have a sort of "download xml every X minutes" and "download xml at boot" flag - QSS (it)

• FEATURE REQUEST: (May 07/07) Move "Download SCR XML" and "Erase Custom SCR" from Preferences to Config
  • Even with configuration locked, the end-user can enter Preferences menu and perform an Erase Custom SCR command. It would be solved if this menu is moved in the configuration menu - QSS (it)

• FEATURE REQUEST: (May 07/07) Add "Lock Keypad" feature
  • It would be useful to have a "Lock Keypad" feature, to prevent end-users entering the menu with the 5-key small keypad - QSS (it)

• FEATURE REQUEST: (May 07/07) Extend idle-screen image from 130 to 131 pixels
  • I noticed that i can only upload 130x64 images, but the screen seems to be 131x64. In fact, if you enable the status bar, upload a white 130x64 image at position (0,0), you will notice that there is a small pixel on top-right of the screen: that is the last pixel of the status bar line, so the screen seems to be 131 pixel large, but if you upload images larger than 130 the xml will be rejected - QSS (it)

• FEATURE REQUEST: (Apr 04/07) SLOWEST SPEED OF BLINKING LED FOR ON HOLD BLF
  • Now that Asterisk 1.4 support Shared Line Appearances, would be nice to support onhold state for BLF, for example blinking the led at slowest speed that in ringing state, so a user can recognize and take the call on hold, like a traditional key system.
  • Now, with notifyhold=yes defined in sip.conf, when you put on hold, the GXP2000 send to asterisk 'Got SIP response 415 "Unacceptable Content-Type"'; the Snom instead support this and the BLF start to blink - Francesco_r (it)

• FEATURE REQUEST: (Feb 27/07) RING TONES
  • Automatic Upload of Idle Screen (not by user request).
  • Upload ring tones with provisioning, not with firmware upgrade, to have different ring tones by phone - Tetard (fr)

• FEATURE REQUEST: (Dec12/06) DTMF
  • Generate DTMF sounds from Multi Purpose Keys if we press them during a call. (Can be used to create custom transfer keys, or to use it for direct access to other PBX features) - lenker (de)

• FEATURE REQUEST: (Oct01/06) Localisation Is Grandstream ever going to provide a way to localise things like dialtones, ringtones, busy tones etc? Not *everyone* lives in the US.- RobH (au)
  • NOTE: Kudos to Grandstream for their implementation of this in 1.1.2.23.- RobH (au)

• FEATURE REQUEST: (Oct11/06) SIP MESSAGE
  • After Grandstream added SIP MESSAGE - support, it would be nice, if they added support for the Content-Dispositon = desktop Header. Background: This Method is implemented on all Snom-Phones and it is very useful for status-information. The sent message appears in the last line of the display and can be deleted by sending an empty message (test via /usr/bin/sipsak -M -O desktop -B "test" -s sip:${USERID}@${PHONEIP} -H ${SERVERIP} -vvv ) - bladerunner

• FEATURE REQUEST: (Oct11/06) TFTP Provisioning
  • Add polling of a generic configuration file (for common settings, like SIP/DNS/NTP server, etc), prior to the specific one (cfgmac).

• FEATURE REQUEST: (Oct5/06) Expansion Units
  • Possibility of adding more than 2 expansion units to the phone.

• FEATURE REQUEST: (Jun8/06) Some useful things
  • Call forward if no answer, after settable timeout function
  • 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 (example: 4 lines-1 call in progress, 1 call waiting & 2 lines left, but busy for incomming calls)
  • set callforward by pressing transfer button, while no call is in progress (instead of dialing *72) and number. It could be for transfer all, while on hook and transfer if busy, while doing it off hook.
  • 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?
  • auto redial function if destination is busy.
  • more ringtones - bad2Dbone

• FEATURE REQUEST: Preloaded ring tones It would be great to have three different ring tones preloaded on the phone, instead of having to upload our own tones to the phones to enable distinctive ring feature. Nothing fancy, just three different simple ring tones. - Kurgan
  • Note: (Jul03/06) 1.1.0.16 comes with three new ring*.bin files in the zip download. Can't say I think they're great, but it's a start. - ninthclowd

• FEATURE REQUEST: (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: (Sept25/06) Agree with datu ! This feature is very common and will arrange echo problem... Best would be that user are able to choose between the 2 mic - flo_turc

• FEATURE REQUEST: Allow the user to dial a phone number without having to take off the handset or pressing the SPEAKER key before. This 'feature' works on all phones (see cell phones or other voip hard phones). It should be possible to type in a number, then pick up the handset or press the speaker button to actually dial. The SEND button would not be needed with this dial procedure. - cheetah
  • NOTE: (Feb22/06) Agreed --Alex
  • NOTE: (Feb22/06) Agreed. This is called On Hook Dialing and would be VERY useful. - Helix
  • NOTE: (Apr/28) Agreed. We installed 30+ GXP 2000 in our company and this feature is urgently missing as people are used to dial on hook and constantly ask me why its not working in this phone. - Datumaster
  • NOTE: (May23/06) Oh man this would be nice! The problem is that there is no Release/Cancel key on the GXP-2000 (A major design flaw IMHO), meaning you would have to use another key(like the MUTE/DEL key) to cancel it if you accidentally hit a key. What if it was possible to use one of the speed dial buttons as a global release key? You could use it as a global "cancel" key as well. You could do the same for any function that the phone supports(i.e. REDIAL or callback), just by adding something like "RELEASE" to the speed dial dropdown menu in the GUI. - ninthclowd
  • NOTE: (June 8/06) Agreed. This feature would be very nice. We have some "very slow dialers" here and even if I set the timeout to 15 seconds (maximum time for GXP-2000 and BT101) it's sometimes not enough. It would be much better to let them dial as slow as they want and then pick up the phone... - graffiti
  • NOTE: (Jun16/06) ditto about the slow dialiers. Not everyone is a 19 year old, 6 point font reader and speed dialer. - Anthony
  • NOTE: (Jun21/06) I would like to be able to type 9 to take the phone off hook like the PBX system I am replacing. - Diver
  • NOTE: (Jul26/06) Definitely high on my personal list of desired features. You can use the Hold key as a cancel button. Ultimately, what would be great is the following: Start typing a phone number. As you type, the numbers display at the top of the screen, (call this the "digit entry line".) Use the "Mute/Del" key to delete, and the left and right arrow to move the cursor left and right in the entered number. After each digit, search for possible matches in the phone book and recently used numbers lists. If one or more matches are found, auto-complete the entry, with the autocompleted digits highlighted, and allow the user to press the Up Arrow button to accept the autocompletion - otherwise, lifting the handset, pressing Speaker, a line button, or the circle button dials only the digits entered. If there are more than one match, autocomplete the first match in the list (sorted in ascending order) and display additional matches in a list below the line on which digits are entered. Allow the user to press the down arrow to enter the list, then use down and up to cycle through matching numbers, and finally to press the circle button to populate a number from the list to the digit entry line. Pressing a number while navigating the list adds to the entered digits and switches focus back to the digit entry line. Pressing Up Arrow at the top of the list should move focus back to the digit entry line. Pressing Hold at any point in this cancels the entire digit entry process and returns to the idle screen. - PusherRobot

• FEATURE REQUEST: (Nov25/05) SEND TEXT. I think I speak for many GXP-2000 users when I ask Grandstream: Please implement SEND TEXT support (and maybe even SEND IMAGE) so we can send custom text to the GXP-2000. Imagine for just a moment what this would mean, especially if the GXP-2000 could be configured to send events on button presses. Phone books could be implemented on a PBX (making them portable) but appear to be on the local phone. Any number of features could be implemented an appear to be local to the phone. This might even make this feature request section get a lot smaller, because a lot of capabilities could be integrated at the PBX level. - thetatag
  • NOTE: (Nov 25/05) I second this. If you could run simple applications on the phone, either with extensions and SEND TEXT/Image or possibly with some simple XML format (ie aastra 480i), that would make this phone a serious business condender, competing with phones that cost 2x as much. Being able to access a centralized directory, even if it's just a dinky SIP or XML script, is a very useful feature. --Helix
  • NOTE:(Jan 12/06) Thirded! :) The large backlit LCD is very pretty, but mostly a useless gimmick right now. Currently the LCD doesn't provide any more useful information than a cheapy 2-row LCD phone. XML browser support like Cisco or Aastra would be a MAJOR win. - bani
  • NOTE:(May 07/06) Fourthed! :) I'm setting up an Asterisk PBX system and using all GXP-2000's. Some of our managers have access to multiple voicemail boxes. Setting up multiple voicemail boxes in the SIP conf for each is fine, but they never know which voicemail box has messages, or how many. I can use SEND TEXT and the PERL or PHP AGI to report how many voicemails are in each box. This would be great to have on the display if they pressed a custom defined speed dial button, for example. - brian

• FEATURE REQUEST: XML/XHTML microbrowser support would be nice, like Snom, Cisco, Aastra and Polycom do it. But even SendText support would be better than nothing. - bani

• FEATURE REQUEST: (Nov23/06) XML Provisioning Now that Grandstream obviously has got some means of XML parsing build into the firmware, an intelligent XML based provisioning scheme aught to be high on the feature wish list. The current scheme with binary files are plain silly and hard to work with.

• FEATURE REQUEST: (Nov25/05) OPEN SOURCE. (I just had to add this one.) If Grandstream were to open source their firmware and allow other private developers to add features and fix bugs (with Asterisk-esque developer oversight), they would probably have a much better product for it. - thetatag
  • NOTE: (Nov25/05) (i agree please OPEN SOURCE) i understand that you likely have segments when are propriatry to other companies (like the Adaptive Digital Echo cancellation) but even if you were to release source code with the propriatry parts removed it would be great as we woul still be able to write code to replace the missing parts and changes could still be rolled into the grandstream firmware if this was your wish... - SoloFlyer
  • NOTE: (Feb06/06) I agree! grandstream sells hardware, not software. aredfox released PA1688 open source and became a much better product because of community contributions. - bani
  • NOTE:(Feb12/05) That may not be possible because of third party licensing requirements, such as the codecs. I wonder how the manufacturer of PA1688 solved this? Maybe other parts such as the user interface still could be opened? -job
  • NOTE: (Feb22/06) Agreed --Alex
  • NOTE: (Jun16/06) Ditto and agreed etc --Anthony

• FEATURE REQUEST: The ability (user-selectable from the web interface or the local phone menus) to avoid taking calls by simply lifting the handset. Suppose you have two incoming calls that are ringing on "line1" and "line2", and would like to take the second call only... I'd like to pick the handset up and get a dial tone, and then press one of the line buttons to take the call on that line, or ignore the ringing calls and place an outgoing call on line three. On busy system this could be useful, because otherwise when the phone rings I am forced to take the ringing call, and cannot do anything else. - kurgan

• FEATURE REQUEST: iLBC or SPEEX codec
  • NOTE: (Feb07/06) Agreed --Helix
  • NOTE: (Feb12/06) That would be very welcome. The only low-bandwidth good-quality codec this phone offers is g.729 which carries a licensing cost at the server. If the processing power is there, having any of these codecs would be extremely useful. - job

• FEATURE REQUEST: Ability to configure the default ring volume (as a user setting please...)
  • NOTE: (Feb16/06) Or at least remember what it was last set to... --SoloFlyer
  • NOTE: (Jun16/06) Good idea. I just included this as a recommendation for solving inaudible pages (see above) - Anthony

• FEATURE REQUEST: I'd like the IP settings to be in Admin config rather in User config
  • NOTE: (Nov??/05) perhaps this can be a config option- restrict IP setting to admin interface? IMHO the main reason for keeping them separate is a user can setup the device as CPE but not change account settings - helix
  • NOTE: (Nov??/05) well... I've put it here because my provider doesn't want to give me the User conf. password because he thinks that I could somehow mix the IP conf. up... - ??
  • NOTE: (Nov??/05) then your provider is a moron. You can set DHCP=yes or an ip address in the TFTP file, which will be auto downloaded to your phone, preventing any change from lasting very long. IP address is widely considered to be a user-side feature, and that prevents you from using speed dials. Admin settings are locked for providers selling the phone as locked CPE or businesses that dont want users messing with the nitty gritty... the IP address *shouldnt* be in admin unless its by an option IMHO. - helix
  • NOTE: (Nov25/05) (i agree your provider is a moron. :) but my preference would for ip address stuff to be under admin, but the option of where to put it would be even better - SoloFlyer

• FEATURE REQUEST: Option to have phone produce a pleasant tone when dialing, rather than DTMF

• FEATURE REQUEST: Configuration option to make a line behave in "immediate" mode — where a number is automatically dialed when the handset is lifted or the line is selected in speakerphone mode. This, especially coupled with Send Text support (see above), would allow a PBX to handle all of the dialing issues if one wanted to do so. It would also allow for these phones to be used to play announcements, or a particular account/line to be connected directly to an overhead paging system. (And the list goes on and on.)
  • NOTE: (Feb07/06) Agreed --Helix

• FEATURE REQUEST: Use a slow steady red blink on line appearances for MWI. If you register more than one account, you don't know which account has voicemail until you try them all and see which one has the stutter!
  • NOTE: (Nov25/05) Another solution, which I would prefer, is to have an indication on the display. It's a big display. Let's use it for something useful. - thetatag
  • NOTE: (Feb07/06) Perhaps have the MWI light blink slowly and the LINE light blinks in unison? -Helix
  • NOTE: (Feb12/06) MWI blinks would be more useful for more line status modes. An indication in the display would be better. Or having the MWI for example blink three times, pause, three blinks etc. for third line. Any (or both) of these indications would be great. - job

• FEATURE REQUEST: Add support for SIP header Alert-Info to trigger custom ringtones, e.g. SIPAddHeader("Alert-Info: ring3"). - bani
  • NOTE: When the phone is ringing, it would be useful to be able to press MUTE (or similar) to suppress ringing on that call. For example, EX-GIRLFRIEND shows up on the caller-id and you don't want to take the call or even listen to the ringing, you should be able to suppress the ringing. The line would continue to blink and could be picked up if you were so inclined. A new call coming in should ring normally, unless it too is suppressed.
  • NOTE: (Feb06/06) if grandstream supported Alert-Info for custom ringtones, you could just make eg a ring3.bin of silence. asterisk could then send SIPAddHeader("Alert-Info: ring3"). I think this is the best solution. - bani

• NOTE: Selection of distinctive ringtones based on multiple numbers, or specific context to distinguish calls originating from PSTN and those coming from internal extensions.
• NOTE: (Dec03/05) On the same note, as well as wildcard matching for rings, it would be nice if the server could specify which of the three rings to use via a SIP header. Ideally the server could specify a url to an audio file ala Snom 360, but that would be much harder to do --Helix
• NOTE: (Feb06/06) I think Alert-Info header is the best method for this. Asterisk can handle this much more flexibly than any phone ever could. It should be asterisk's job to decide which ringtone to play. - bani
• NOTE: (Feb06/06) I second this. We need an alert-info header for distinctive ring, and four factory-default decent ringtones, too. - kurgan
• NOTE: (Feb06/06) Distinctive ringtone selection with wildcard or SIP header. Best implemented through SIP header? Or with wildcards in the advanced settings..(an X or a . like in asterisk?) - stoffel
• NOTE: (Feb12/06) This phone is also useful for home deployments where you hardly install a PBX (for which a softphone would be more cost-effective), so having the option in the phone book be a nice bonus. I also second the SIP header proposal but it should be on a per line basis as the auto answer header is. - job
• NOTE: (Jun16/06) I think this should be considered a major bug. Anyone up for raising this to a major fix? (my neck is not long enough yet). - Anthony

• FEATURE REQUEST: This isn't directly related to the GXP firmware, but I'd like GS to document the script format used by their ringtones. The GXP comes with a ring file that will ring once, then say "You have a call from" and read off the caller ID number. With documentation of this data format, it would be possible to write custom ring tones that do the same thing but change the ring sound and the prompts. - Helix

• FEATURE REQUEST: GS please supply a dozen of ready made decent ringtones to be able to use these phones on desks close by. Realise that ringtone converters exist but would gladly avoid the audio format mumbo jumbo to have a phone ring differently! - bengrech

• FEATURE REQUEST: Make the GXP respond to a check-sync NOTIFY event and reboot or re-tftp. The 'grandstream reboot script' is a hack. - Helix

• FEATURE REQUEST: Phone should accept NTP settings via DHCP, eg option ntp-servers and option time-offset from ISC DHCP. - bani
  • NOTE:(Feb08/06) The option ntp-servers works for me in firmware version 1.0.1.9 using ISC DHCP. Is this broken in 1.0.2.x? bklang
  • NOTE:(Feb08/06) Are you sure? I don't think this ever worked, even in 1.0.1.9. The firmware comes pre-configured with pre-set ntp time server. To verify that it is actually working in 1.0.1.9 you will want to blank out the ntp server in the advanced settings tab, update and then reboot. See if it actually gets your ntp server from dhcp. - bani
  • NOTE:(Feb15/06) I was wrong; it does NOT work in 1.0.1.9 (thanks bani for pointing that out). I agree this would be a very useful feature especially for heavily firewalled environments. - bklang

• FEATURE REQUEST: The ability to call a SIP address directly (by SIP-URL). - job

• FEATURE REQUEST: Extend BLF support to show more extension states such as ringing. The Snom phones do this very well. - job

• FEATURE REQUEST: Make phonebook and any useful menu item from the display-GUI available in the web-backend please. - PaulK

• FEATURE REQUEST: Storing/retrieving phone book with LDAP would be very welcome for enterprise use. - job

• FEATURE REQUEST: when someone calls, and the number is known from the phonebook, display the name as well as the number. Keep in mind, someone calling from 5 is the same as 098335455 locally for germany. - PaulK
  • NOTE: (Feb06/06) Couldn't asterisk already do this by changing the caller id string? Eg Set(CALLERID(name)="Somename") It's a bit more work, but I think the proper place to do this is asterisk. - bani
  • NOTE: (Feb12/06) The phone book is local to the phone. It cannot be handled by the PBX, except for central phone books which already work as you propose. Under no circumstances should features be asterisk-specific unless there is an extremely good reason. - job
  • NOTE: (Feb12/06) Asking grandstream to hardcode the local calling rules of every city on earth is too much. These kind of special case translations belong in asterisk not the phone. - bani
  • NOTE: (Feb18/06) Agreed the phone should not try to be smart about the numbers. But if the number exists in the local phone book, the phone should present the corresponding name as well. That would be a lot more useful than not doing it at all. I also want to stress that nothing "belongs" in asterisk specifically, it is important to follow standards so the phones are useful with any SIP proxy/PBX. - job

• FEATURE REQUEST: Option to define TFTP 'Firmware Server Path' and 'Config Server Path' via DHCP. option bootfile-name and option tftp-server-name in ISC DHCP. - bani

• FEATURE REQUEST: Option to enable/disable sidetone and adjust sidetone volume. - bani

• FEATURE REQUEST: Extensions number (account 1) should be on display, plus option to turn off showing ip address (users dont need to know it). - zalink
  • NOTE: (Feb12/06) Agreed. Registered username(s) is often much more interesting than phone IP. The display is large so all active usernames and their respective status could fit there. - job

• FEATURE REQUEST: Missed calls should not take over the whole display. Should just have a small message in top right that says n missed calls. - zalink

• FEATURE REQUEST: When on a speakerphone call, Mute button should only mute microphone and leave speakerphone still active. This is especially useful for speakerphone on hold or listening to voicemail - bklang
  • NOTE:(Feb15/06) This seems to be the case after switching to the GSM codec. Is this possible? Only true for g.711u? - bklang

• FEATURE REQUEST: Show current time in the phone status page. There is no REMOTE way to know if the phone has correctly received the time via NTP. - SamL
  • NOTE: (Feb12/06) Also line account status, preferrably with time stamps when the events happened, would be nice. -job
  • NOTE: (Feb12/06) Related: Phone should try to get time via NTP every few minutes if it fails during boot. It seems phone only tries to get the time once, and if it doesn't, it doesn't try again until next reboot. -SamL

• FEATURE REQUEST: (Feb11/06) Pressing Transfer + one of the speed dial / BLF buttons should do a blind transfer to that speed dial extension. - Ted
  • NOTE: (Jun08/06) AGREED! This only makes sense. Makes memorizing extentions a thing of the past! -NJTaz76

• FEATURE REQUEST: (Feb11/06) It would help to make the speed dial buttons usable during a call. They could then be used for special PBX features like call parking. Ideally, in the setup, have two numbers: what to dial if already in a call, and what to dial when using it to make a new call/transfer. Then I can press a button in the call to park the call while in the call. Then use the same button to pick up a parked call. - Ted
  • AGREED. I would like to use one SD button for a door-phone application but cannot use it while the phone is off-hook. - Anthony

• FEATURE REQUEST: (Feb11/06) Support simple plaintext or xml configuration files, without having to use the grandstream java configuration tool. - bani
  • NOTE: (Feb12/06) At least the file and checksum format should be made public. The third party market would love to get native support for provisioning of these phones. Grandstream only loses market share by trying to keep this secret. -job
  • NOTE: (Feb12/06) I think overall the phone needs to be more 'open'- GS should take a hint from AAstra (they provide tons of documentation on how to provision, configure, and write XML apps free for everyone). I think possibly GS wants to keep things a little under wraps so they can sell GAPS licenses (my observation/theory, not spreading accusations). If this is the case, I would at least suggest they consider the possibility that opening their formats and maturing their firmware will result in far more HW sales to the Asterisk community (especially small/medium integrators and hobbyists) than will be lost by large carriers not being locked into using GAPS. FWIW, I fall into both of the above categories and i found the 'grandstream config tool' to be a PITA... -Helix
  • NOTE: (Feb12/06) I don't mind them wanting to use a special binary format. I do however think they'd be much better off if they just publicly documented the format, since then conversion tools can be written in any language, including ones that don't take a full 5 seconds of CPU time to process a text file (i.e. every language other than Java). Though needing to use the tool doesn't particularly inhibit configuration via PHP or XML. PHP can generate the config file, and than then call the java app to compile that file into the binary config for TFTP. Or an XML file could be generated, and a simple XSLT script could convert the XML to Grandstream's text config file format and then run the config tool on that. In fact, you could even use PHP to connect to the phone's web interface, collect current values (via XPath querying form input on the returned page's DOM, and extracting the mappings of PXXX -> value). And one could also use the phone's web interface from a PHP script to update new values. -Ted
  • NOTE: (Feb12/06) What purpose does it serve to have encrypted config files which can only be generated by a special tool? Maybe if you want to make it possible for ITSPs to lock down devices to a single provider (and thus lock out asterisk), but this seems pretty dumb for a device which has explicit asterisk support ("asterisk BLF" in the menus). The binary format is annoying and serves no purpose other than making it more cumbersome to integrate gxp2000 with auto-provisioning tools. It means grandstream is the odd man out (compared to polycom, cisco, aastra, linksys, snom, and just about everyone else on the planet who has open provisioning formats). additional: having to execute external closed-source binary programs opens up a whole new can of worms security-wise. if you can keep all functionality selfcontained within your cgi, things are extremely simple to lock down. as soon as you have to start doing system()/exec() etc, the floodgates are open. - bani
  • NOTE: (Feb12/06) I agree with Ted, my main complaint is not the file format itself but the lack of openness. Using AAstra as an example again, they use a XML based file, but allow it to be encrypted so the phone can be locked by an ITSP (they provide a tool to generate a keypair and put the decrypt key into the phone). The difference between AAstra and GS is that GS seems to be (possibly) keeping the file format under wraps so they can make money selling that tool, which they call GAPS (Grandstream Automated Provisioning System). GAPS can also remote-reboot a phone via an undocumented means (or so I've read). While I dislike locked HW as a concept, I am not against it for the purposes of this discussion. My complaint is that useful features (remote-boot phones, generate our own config files using 3rd party tools) are *possibly* being withheld so GS can sell more GAPS licenses (why would an ITSP pay $lots for GAPS when they can download some dude's config script for free?). Keep in mind I am *NOT* accusing anybody of anything, just theorizing. Another possibility is that the binary format is kept under wraps because it is written to flash with little inspection or modification by the phone. I dont care if they make the configs XML based (that would be preferred) or document the existing format, I just care about the end result which would be a much more useful and less clumsy way of centrally managing GS phones, which would make me more likely to use the GXP for professional deployments. -Helix
  • NOTE: (Feb15/06) While I think an open configuration format would be nice, I would like the closed format kept too. I autoconfigure my phones across the public Internet and if someone were to unscramble one of the files, they would find out the SIP account passwords. It's true that I could get the phone users to enter their passwords manually but that takes me one step away from a totally autoconfiguring system. I do realise that these files could probably be reverse engineered without too much effort but I doubt anyone would want to abuse our system that much. As for the practically of using such files, I set up a simple shell script that makes generating them quite simple for us. - Chewi
  • NOTE: (Feb16/06) Maybe im missing something... but it is possible to create config files which are accepted by gxp-200's but the passwords dont get encrypted they just do the checksum, it was reverse engineered by a guy on the digium mailing lists who wrote a perl script to make the files... it has worked for every firmware version i have tested it on (up to 1.0.1.13) it is attached here --SoloFlyer
  • NOTE: (Feb16/06) Hmmm, thats actually pretty useful, i'll see if it helps (I wasn't aware such a thing existed, I had looked for one a while ago with no luck). Thanks!. However Chewi- an obfuscated config file and a secure config file are two VERY different things. GXP cfg file is somewhat obfuscated, but not secure. Unless the phones are using some kind of cryptography, then all you have to do is figure out how exactly the password fields are being scrambled, and you can unscramble them all. (i'm talking theoretically btw). This perhaps is why they have not documented the full file format. In contrast, AAstra uses open XML for everything, but when you need security, you use their tool to generate a real crypto key to load onto the phone before shipping it to the user. Their tool will then use the same key to crypt the XML file, resulting in crypted output which is totally useless unless you know the key (which the phone does). IMHO this is a more elegant solution. However since it seems (from this script at least) that the main purpose of the cfg format is to provide integrity (checksums/etc) I have much less of a problem with it. While I'd still like XML, or even real GS docs on the format, this looks to be much more workable now. - Helix
  • NOTE:(Jan23/07) The file attached here posted feb16/06 works for 1.1.1.14, can someone please give me a hand converting the pack/unpack functions to PHP? - jamesb63

• FEATURE REQUEST: (Feb14/06) My users don't like seeing missed call logs for the ring groups of which they are members; they only like seeing missed calls specifically for their extension (not the generic ring group). Please add support to ignore missed calls to ring groups by examining a Caller ID prefix. In Asterisk, you can append a prefix to the Caller ID of all calls to a specific Ring Group. For instance, we prefix "Sales:" and "Support:" to the Caller ID of calls into those ring groups. By parsing the Caller ID, you could easily filter which calls make it to the Missed Calls list. - gammacoder
  • NOTE: (Feb14/06) Like so many other things, this could be handled if the GXP-2000 used sip headers (e.g., X-GXP2000-MissedCall-Logging: no) to control a lot more of its functionality. The phone should probably give the option "Support X-GXP2000 SIP Headers?" for situations where the remote connection is not trusted or authoritative. For most PBX implementations, however, having those SIP headers available for control would grant an enormous amount of control to the administrators to tweak the system to its highest potential. - thetatag

• FEATURE REQUEST: (Feb15/06) UPnP Support - I have set up my company's phones to autoconfigure. The only thing that prevents us from complete autoconfiguration is the fact that anyone behind a NAT router has to manually forward UDP port 5004 to their phone. If this could be done automatically using UPnP, that would be great. I'm not sure what UPnP is capable of but if it is able to query the router about what ports are available, it would be possible to have several phones behind the same NAT router without any extra configuration being necessary. - Chewi

• FEATURE REQUEST: (Feb15/06) DHCP Hostname support - Adding the ability to configure a Hostname on the phone which will be sent in the DHCP Request packets will help with dynamic DNS. - bklang

• FEATURE REQUEST: (Feb19/06) Settings should take effect even without a reboot. Some settings may still need a reboot - a reboot should be proposed then.- PaulK
  • NOTE: (Feb20/06) Many settings do take effect as soon as they are changed and do not require a reboot (Speed dials for example)... but it would be nice for the phone to only suggest reboot when actually required... --SoloFlyer

• FEATURE REQUEST: i would like to be able to control the AGC gain from the admin page.-Alex

• FEATURE REQUEST: (Feb28/06) I would like to be able to hit "mute/del",or something when the phone rings to make the phone ring silent.-mtryfoss

• FEATURE REQUEST: (Mar18/06) Multilanguage support - For all us not english speaking, I suggest getting language out of the main firmware file and let it be downloadable like ringtone files. If I could adjust those files myself, it woulf be great, too (say like GNU multilanguage support). - KampfCaspar

• FEATURE REQUEST: (Mar18/06) Arrow key functions - Arrow keys should ALWAYS lead to some interesting information, eg. Arrow-Up is ALWAYS the missed calls list (even if empty), Arrow-Down ist ALWAYS the Phonebook... As an added extra, the function might be configurable. - KampfCaspar

• FEATURE REQUEST: (Mar18/06) "Send" = redial -"Send" Key in idle state should get one to the dialled numbers list. Upto now (1.0.2.13) I seem to only be able to call the LAST dialled number directly. - KampfCaspar

• FEATURE REQUEST: (Mar18/06) Mute/Del = Escape - In whatever menu or function I am, I want a single "get me out one level" key. I think the Mute/Del key would be fine for that... Even if it gets a little crowded if DND in idle mode and stop ringing in ringing mode are added, too.... - KampfCaspar

• FEATURE REQUEST: (Mar18/06) Transfer ringing call - I would like to transfer a call BEFORE accepting it. Like the phone rings, I press TRNF and a speed dial - gone. A redirect to voice mail would be great, too (TRNF & MSG). - KampfCaspar

• FEATURE REQUEST: (Mar18/06) Idle display configurable - Depending on usage, the most important idle information is different. Let it be configurable if you want the date as prominently displayed (or rather all date/time in the topmost line), if you even need a bold "username/number" display or rather some information about the four lines or other users (like the boss knows his name but wants to know if the secretary is online), etc. As an added bonus, let it be some XHTML/WAP display (like suggested above) - KampfCaspar

• FEATURE REQUEST: (Mar18/06) Action Urls - Like with SNOM - probably even enhanced - would solve a lot of issues. - KampfCaspar

• FEATURE REQUEST: (Mar20/06) BLF with CLIP - Ability to display calling number information on a blf key (secretary get's number calling boss). - KampfCaspar

• FEATURE REQUEST: (Mar24/06) Phone Book Load via TFTP - As a feature for small-to-medium deployments, why not have the option to populate each phone's internal phone book from an easily editable text file (or whatever) via TFTP on reboot. This would seem to be a reasonable compromise for the small business/serious home user/hobbyist between the pain of editing each phone individually and running a full-blown centralised database (assuming we get support for that one day). It's a lot quicker and easier to reboot 6+ phones than edit them! - Channel-Two
  • NOTE: (Apr17/06) Agreed - it would be great to have a way to upload a user-editable file of numbers (by TFTP or some other method); it's a pain to have to key them in, and even worse to have to update each phone individually --ADW

• FEATURE REQUEST: (Mar29/06) Automatic Daylight Saving switching - Twice a year I have to update lots of phones to change daylight saving settings from YES to NO and back. It would be great to have the phone do this by itself. - Kurgan

• FEATURE REQUEST: (Apr17/06) Dial Plan enhancements - I have set up a dial plan on my Sipura SPA-3000 such that the local area code is automatically inserted for calls to my local area (here, any number starting with 7 or 8 is a local number). This makes the conversion from PSTN more transparent for my users who are used to making local calls without having to dial the area code. Is there any way to do this with the GXP-2000? It would make the phone more acceptable if it were possible. - ADW

• FEATURE REQUEST: (Apr28/06) display messages in German language - my German customers are not willing to buy a phone which has no German user interface, at least the basic displays should be adjustable in other languages. - Datumaster

• FEATURE REQUEST: (May1/06) On-hook answer button. Option for configure button for on-hook answering for working with RJ9 modular connector headphones. - cupotka

• FEATURE REQUEST: (May 3/06) Use DHCP option 88 to set DST. This is the right way of getting the DST rules (when does time change, and by how many hours) to display proper time. Better than faking a wrong timezone calculated by adding DST to local timezone and setting it with DHCP option 2. Using both options 2 and 88 would allow for a complete and proper time setting for both timezone and DST rules.- Kurgan

• FEATURE REQUEST: (June 8/06) It woud be nice to have the possibility to switch the LEDs on the speed dial keys (maybe as an extension to the send text feature requested above - a special formatted text should do the trick). This could be used e.g. from Asterisk to signal whether another phone is ringing (LED blinking) or busy (LED on). - graffiti
  • NOTE: (Jun08/06) This is already possible. As of firmware version 1.0.1.13 (I think, it is very old) the speed dial keys (basic setup page) have a type option for "Asterisk BLF". If you set this as well as subscribecontext= in sip.conf and hint exten's in the subscribe context, the LEDs will provide presence for monitored extensions. Off=not on phone, button is a speed dial. Blinking=monitored extension is ringing, pushing button will dial its exten with ** before it (so you can set **exten to be pickup). On = user is on the phone, button does nothing. - Helix
  • NOTE: (Jun08/06) Thanks for info. I've never realized what BLF really means. - graffiti

• FEATURE REQUEST:(Jun15/06) SRTP support Maybe I'll get flamed for this, but privacy is important... especially in a business environment. - grsch

• FEATURE REQUEST:(Jun30/06) BLF InUse option to light only when line is "InUse" vs "Unavaliable" (not registered) - peter

• FEATURE REQUEST:(?????/??) One button Transfer It would be great if you had e.g. You pick up a call — click on an extension — call is transfered. - joekane

• FEATURE REQUEST:(?????/??) Extension loading Would it be possible when the phone is loaded all the extension lights come on at the same time, instead one at a time. Thanks for the improvements so far. - joekane

• FEATURE REQUEST:(Oct19/06) SIPS support SRTP makes not too much sense without SIPS ;) (thx nevertheless for the implementation of SRTP so far) -kodomo

• FEATURE REQUEST:(Oct18/06) Transfer Tone It would be nice if during an attended transfer there was a tone/beep/note played to the person that the call is being transfered to, so that they know when the transfer has been completed -SoloFlyer

Download Firmware

• - Beta Build 1.1.0.11
• - Unofficial Internal Build of 1.1.0.1 "(Fixed Silent ringing also DTMF during a call is kept on the screen)"
• - Unofficial Internal Build of 1.0.2.13 (Distribution for testing sanctioned by Grandstream)
• - Unofficial Internal Build of 1.0.2.8
• - Unofficial Internal Build of 1.0.2.6
• - Unofficial Internal Build of 1.0.1.13
• http://www.grandstream.com/BETATEST/GXP2000_BT200/ - Latest Beta
• http://www.grandstream.com/DOWNLOAD/FIRMWARE/GXP-2000_BT200/ - Latest "Stable"
• http://www.grandstream.com/y-firmware.htm - English Firmware Links
• http://www.grandstream.com.cn/y-firmware.htm - Chinese Firmware Links

Instructions to Upgrade a GXP-2000 to Firmware version 1.0.2.3 using Linux - gammacoder

• On your Linux Box (Make sure you have a tftp server installed, the package for Fedora/Redhat is 'tftp-server')
  • cd /tftpboot (or your tftp server root directory)
  • wget http://www.grandstream.com/BETATEST/GXP2000/Release_1.0.2.3_GXP2000.zip
  • unzip Release_1.0.2.3_GXP2000.zip
  • /usr/sbin/tcpdump port tftp (to watch tftp requests from the phone)

• Point your webserver to http://ip.add.dr.ess from the phone's front panel
  • Login with the administrator password
  • Advanced Settings: set Firmware Upgrade: Via TFTP Server and set the IP address to your TFTP server
  • Click Update
  • Click Reboot

• Watch tcpdump, you'll see requests for
  • boot55.bin
    • If it hangs here for more than 10 seconds, pull power and reboot.
    • On the phone's Web Interface, under Status, you'll have Bootloader-- 1.97.1.99
    • Reboot the phone from the web interface one more time
  • boot55a.bin
  • boot55a.bin (again)
  • gxp2000a.bin
  • cfgxxxxxxxxxxxx (where xxxxxxxxxxxx is the MAC address)
  • cfg.txt

• The phone boots up, then runs through the new provisioning code. In tcpdump, you'll see:o
  • cfgxxxxxxxxxxxx (again)
  • boot55a.bin
  • gxp2000a.bin
  • ring1.bin
  • ring2.bin
  • ring3.bin

GXP-2000 central updating TFTP server (GXP-2000 firmware v1.0.1.9)

• The first time you have to enter the TFTP server-IP in your phone through the web interface, as of then it will always look for the cfg file on the TFTP server.
• On grandstream.com you find the Configuration Tool, you need this to convert the config file to the correct format. A config template can be found on that page also.
• My cfg file is called cfg<mac-addr> and is placed in the tftp dir. If you have multiple phones, you need multiple cfg files.
• Here is what I have changed from the default template:

• After creating your config file, you need to convert it with the Configuration Tool and make them available on your TFTP server. (filename syntax: cfg000b8201XXXX)

Update and Modify a GXP-2000 from the command line

• Here is a modified perl script gsutil.txt that I believe works with firmware up to and including 1.1.0.16. It has a .txt ending since the wiki does not accept .pl

• (Jul 19/06) It works with 1.1.1.7 also but does not have the new bells and whistles for the extension unit, or some info to make adjusted parameters such as BLF presence work.

• (Jul 26/06) Here is a modified gsutil.txt with the added info needed for the extension unit and firmware 1.1.1.7

• Firmware Version 1.1.1.7 -ninthclowd

• Firmware Version 1.0.1.12

• Firmware Version 1.0.2.13

• These were emailed to me directly from Grandstream some months ago. Others may find these useful.

• I would really like to make each of the firmware version configuration templates files available here, but I do not have them. Can anyone provide more?

Custom Ring Tones

Hardware Notes (GXP-2000)

Hardware revisions

• 16 Sep 2006: Some newer units that I have bought recently have dual color (green/red) LEDs for line buttons and speed dial/BLF buttons, but it seems that current firmware does not use this feature at all. In these units LEDs glow green at boot, then always red during normal operation. These units came with firmware 1.1.0.14 preloaded. - Kurgan
  • The 1.1.0.x series firmware does not utilise the green LEDs, however, the 1.1.1.x firmware does. - RobH (au)
  • How? I only see them green when configured as BLF, and then red or blinking red when line is in use/ringing. It seems that the only difference is that leds are green instead of not lit, when line is not in use. What's the purpose of the green light? Kurgan

Hardware faults and problems:

Build your own headset adapter

Buy a headset adapter for the GXP-2000

Headset Adapters now included in GXP-2000 packaging.

• Have to agree: I have often found that moving GXP2000s causes them to reboot due to a 'loose' power connector. - Linker3000

XML Phonebook

XML Idle Screen

Label Template

Buying a GXP-2000 (Alphabetical by country then domain name)

• USA - 888VoipStore - Same Day Shipping, In-Stock Now! Shop and Save at 888VoipStore! Grandstream GXP-2000 $78.95
• USA - Atacomm - Lowest pricing on Grandstream, Volume Discounts Online - Free Ground Shipping & Support
• USA - DigiumCards.com
• USA - discountvoipoutlet.com
• USA - powersystemsdirect.com Works very well with our Talkswitch SOHO/SMB Phone Systems, Platinum Dealer
• USA - Telephony Depot
• USA - ThinkBright
• USA - Voiplink.com (Located on West Coast, Global Distribution)
• USA - VoipSupply GXP-2000 only $79.99
• USA - VoIPPabx.com free ground shipping
• Australia - Australian Technology Partnerships
• Australia - OzVoIP.comaaa
• Australia - Techtopia
• Australia - voipshop.com.au
• Belgium - http://shop.packetnet.be,
• Belgium - VoIPsolutions (resellers prices avaible) 83 EUR ; Delivery all of Europe
• Belgium - WeePee VoIP in combination with a WeePee VoIP Subscription
• Canada - AtlasVoice
• Canada - www.grandstream.ca
• Canada - netVOICE communications
• Canada - VoIPPabx.com free shipping
• Colombia - Gecko Networks
• Costa Rica - PBX Virtual
• Czech Republic - Daktela - VoipObchod
• Czech Republic - VoIPEX
• Egypt - ATSINT (payment in US or EGP)
• Europe (All) - AccesIP - France & Europe
• Europe (All) - beroNet VoIP Shop (payment in EUR)
• Europe (All) - Direct Centrex (payment in EUR, resellers prices avaible)
• Europe (All) - http://www.norskit.com/nit/default.asp?PageNo=DEFAULT&DeepLink=PB10000D%2CHardware%2CTELEK%2DVOIP110019633 90 euros 16% German VAT inc.
• Europe (All) - iTalk (payment in EUR)
• Europe - MyPhoneCall Europe's largest distributor of VoIP products. (payment in GBP or EUR)
• Europe (All) - Packetnet Shop (payment in EUR)
• Europe (All) - VoIPsolutions At 83 EUR; (resellers prices avaible)
• Europe (All) - x9media ip communications (Germany, Düsseldorf)
• Ireland - iTalk
• Israel -http://www.tikalnetworks.com
• Italy - VoipShop.it
• Mexico - Intruder Consulting S.A. de C.V.
• Mexico - Telecomunicaciones Abiertas de Mexico SA de CV
• Middle East - ATSINT (payment in US)
• Netherlands - Voipwereld shop
• New Zealand - http://nicegear.co.nz
• Norway - NovaOffice
• Norway - Voipbutikken.no
• Norway - TeleVoIP
• Poland - easyCALL
• Portugal - Voxsys, RSCE (Distribuidor GrandStream)
• Slovenia - ORG. TEND â?? evolucija informacijskih sistemov
• Spain - Avanzada 7
• Spain - Capa Tres Soluciones Tecnologicas S.L.
• Sweden - inWarehouse
• Venezuela Global IP Systems de Venezuela C.A.
• Worldwide - Atacomm (Largest Stocking Grandstream Distributor in North America)
• Worldwide - Global IP Systems
• Worldwide - iFAX Solutions
• Worldwide - Telephonyware
• Worldwide - The VoIP Connection Certified Grandstream Partner. HUGE INVENTORY!
• UK & Worldwide - VoIPon Solutions
• Worldwide - DigiumCards.com
• Worldwide - VoIPPabx.com
• Worldwide - Voxilla Store - Grandstream GXP-2000
• Worldwide - VoIP-Stock.com
• Ukraine - Wildix in Ukraine

Reviews:

• AsteriskGuru Review
• VOIPSpeak Review
• VoIP User Review
• Outdated review of the GXP-2000 at aussievoip.com.au
• Jacken's Blog Review

See Also:

• Grandstream
• Grandstream GXP-2000 - Solving Echo Problems
• Grandstream GXP-2000 Configuration Wizard
• Grandstream Support Forum
• Asterisk Intercom/Paging with Grandsteam - BEZ(zktech)
• GXP-2000 Firmware Hacking
• GXP-2000 Legacy Firmware Notes
• Tutorial on using the GXP-2000 with Asterisk
• How to Configure BLF with Grandstream GXP-2000 and Asterisk@Home - Jacken
• Outdated list of GXP-2000 bugs at aussievoip.com.au
• GXP-2000 Disaster Recovery
• GXP-2000 XML Phonebook
• GXP-2000 XML Idle Screen
• Grandstream Configuration Tool - contains info on configuration file format
• Autoprovisioning Equipments GXP2000/PAP2