Grandstream GXP2000 Firmware Archives

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Firmware Notes ("Beta"

(Apr12/07): The file is no longer on the Grandstream BETATEST site. - Shane Steinbeck
This firmware version suffers from stability issues. There have been many complaints of phones locking up. csnyder
This firmware version has major audio quality issues. Don't use this firmware. ebeheler

(Mar20/07): Firmware, Release Notes, Language pack.

WARNING! You cannot downgrade from this firmware.

Changelog (
Build 03/15/2007
  • Fixed attended transfer will fail (as transferee) if Contact header come after Refer header in the REFER request New Features/Changes:
  • Default DST rule changed from "4,1,7,2,0;10,-1,7,2,0;60" to "3,2,7,2,0:11,1,7,2,0;60" in compliance with the U.S. Federal Law passed in Aug 2005.
  • The 403 message incorrectly spelled "alloweded", change to "allowed"
  • In MENU-Preference: "Do NOT Disturb", changed to "Do Not Disturb"
  • Copyright string at the bottom of HTML pages changed from (2004-2006) to (2004- 2007)
  • The DST rule change above will ONLY occur after a factory reset
  • If the GXP-2000 EXT does not start correctly immediately after the upgrade-completion bootup, please power cycle the GXP-2000
  • Fixed under VLAN mode, we send malformed ARP responses
  • Fixed we crash when we do DHCP renew during a call under certain scenarios
  • Fixed we increment UDP version for session-timer refresh reINVITE where session information did not change
  • Changed when we receive an reINVITE without SDP, default to sendrecv in offer (200/SDP) regardless of current RTP state
  • Added support for changing local SIP port on registration failure
  • Added configurable registration backoff interval, P138/471/571/671 in seconds (1-3600, default 20)
  • Changed iLBC default payload type back to 97 (changed to 99 in; note that the actual payload type is not changed

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.3.2.pdf"

Bugs / Tweaks (

Guys I'm sorry I started that flame thread and so I erased all of it and everything that spawned from it. My apologies to anyone who had actual bugs listed. I'm going to try to get some help from Grandstream and if I learn anything that would help you I'll post it. In the future, if you post a comment with a bug, please list your HW version. - ninthclowd

  • 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.
The same problem come when line 3 is ringing and we quickly select this after an attendant transfer from line 1 to 2.
Phone working fine if we wait 1 sec before select line button 3 (until a line led turn on and headset make tone) . - jak
  • 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?" 01/30/2007 - Not yet available for public use):

Changelog (
  • Fixed iLBC default payload type incorrectly labeled as 97 (should be 99)
  • Fixed we freeze if the domain string in WWW-Authenticate contains semicolon
  • Fixed we incorrectly use To-tag in REGISTER
  • Fixed GXP-2000 WEB UI EXT1 page translation incomplete (half of the page has "User ID" in English regardless of language pack)
  • Fixed under VLAN mode, we send malformed ARP responses, shifted by 4 bytes
  • Fixed we do not untag incoming VLAN packets correctly
  • Fixed iLBC frame size/payload type options missing in WEB UI
  • Fixed DHCP options 2, 42, 66 not working
  • Fixed DHCP option 66 does not handle path correctly
  • Fixed NTP does not work when NTP server is in IP address form

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.2.27.pdf" file included with the firmware. Dunno if it is legal to attach the firmware zip here, but if it is legal and you need it please let me know and i'll post it here. - QSS (it)

  • 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" - Not yet available for public use):

  • VLAN support functions properly again

Above information taken from personal testing
Question: Is the router functionality also reactivated?
Question: Any timescales on this? The echo and de-registration issues have crippled my userbase.
  • * 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', despite what some of the notes say. This does not have the hiss on my units.
Grandstrem seem to have released the next firmware (19th March).

Firmware Notes ("Beta"

(Dec18/06): Currently available from Grandstream BETATEST site, Firmware, Release Notes, Language pack.

Changelog (
Build 1/9/2007
� Fixed the �hissing� noise coming from the other parties handset
� Fixed VLAN not working
� Fixed display phonebook entry name as caller ID not working correctly
� Fixed We always use the firmware server in the HTTP host header
� Fixed iLBC bad audio quality
� Fixed GXP-2000 incorrectly performed consultative transfer when you switch line during a blind transfer
� Fixed GXP-2000 results in one way audio when a second incoming call is not answered while the call is on hold
� Fixed we will not register any account if STUN is down or misconfigured
� Fixed if network is down-then-up STUN IP checking gets fired multiple series causing many STUN queries
� Restructured STUN/Registration to simplify account registration management
� Fixed echo in 3WC problem reported in
� Fixed GXP-2000 under 3WC, second call info not displayed correctly
� Added support for BT-200 onhook-threshold.
� Added customizable delayed call forward wait time. Provision parameter P139/P470/570/670, default is 20 seconds (as is previously), allowed value 1-120; invalid values ignored.
� Added support for BT-200 delete called/caller entries via MUTE/DEL key
� Changed NTP will retry 3 times if it receives no response from NTP server; after that it will retry it after 1 minute. This also fixed the NTP problem reported on the wiki site.
� Fixed we do not encode "#" in outgoing INVITE To URI

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.2.25.pdf"

Bugs / Tweaks (

  • 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, 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 and only since firmware version 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 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 'stable' release. I wouldn't mind so much if 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 ( 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 <= - 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"

(Dec18/06): Currently available from Grandstream BETATEST site, Firmware, Release Notes,
Language pack.

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: Previously I had read that you cannot downgrade if you install this update. Well I managed to do it via TFTP, dunno if it was a fluke or not just thought yall might want to know. - ninthclowd

NOTE: (Jul04/07) I Have started receiving phones from grandstream firmware on it from factory, since this is still officially a beta firmware I found that a bit strange, but I have been able to downgrade the firmware on the phones to using HTTP and about 6 or so reboots... I guess it wasnt a fluke ninthclowd - SoloFlyer

Changelog (
Build 12/15/2006
� Fixed a short buzz is heard when TRANSFER completes
� Fixed you can still enter GXP-2000 MENU when the phone is ringing
� Optimize the speakerphone performance of the GXP2000 and BT200 to a 6 ft range. Implemented with a fixed AGC and a rough enhanced VAD based on a 5/16 frame buffering
� Added simple noise suppression for non speaking parties
� Fixed under Broadsoft mode we will not send INVITE if SIP Server/Proxy is in IP address form
� Implemented resuming call when CONFERENCE key is pressed again
� Implemented resuming call when TRANSFER key is pressed again
� Fixed GXP-2000 line key LED will become inaccessible
� Changed syslog or web UI status page for MAC address: separated by colons and in uppercase
� Added call establishment STUN queries to use event callback when response arrives, this reduces the brief delay when making and receiving calls when STUN is configured thus improving call experience
� Added comments on WebUI for Account Name display support for BT-200
� Fixed GXP-2000 WebUI EXT1/EXT2 pages does now contain "eventlist BLF" as option, EXT2 page Key 65 display as "EXT Key 65: 65:", EXT2 page wording "UserID:" missing for Key 71-112
� Take Ring Tone out of Call Progress Tones section and added syntax description
� Fixed some factory blank-LCD problem caused by GUI library not initialized correctly
� Redesigned mic and AGC/VAD changes
� Fixed handset/headset echo issues
� Fixed dial tone click and garbled dial tone in Broadvoice test
� Changed ringer volume gain from analog to digital scaling and increased max ringer volume
� Fixed G.723 on BT-200
� Fixed we allow HOLD to a call with early media
� Fixed under Broadsoft mode we will not register if SIP Server/Proxy is in IP address form
� Fixed we display SRTP error messages when 488 is received
� Added "P-Asserted-Identity" header for anonymous calls by Privacy header under Broadsoft mode
� Fixed BLF does not work with GXP-2000 EXT broken during eventlist implementation
� Fixed GXP-2000 renders bitmaps incorrectly when the encoded bmp string contains CRLF in it
� Fixed volume cannot be adjusted for HW 0.4
� Fixed 3WC audio degrades when G723 6.3k is used
� Changed: we will NOT challenge reboot NOTIFY with 401 and accept it with 200 when SIP Authenticate ID is not configured.
� Fixed we do not use anonymous URI when making anonymous call using Privacy header as per RFC3325
� Fixed we sent RTP under MUTE in PCMU (regardless of the actual payload type in use) causing a short audible sound on remote end. An invalid RTP is sent instead which will be dropped but still keeping the NAT binding alive
� Fixed we do not sent RTP keep alive under HOLD
� Added tone analysis (disabled)
� Added support for save call history entries to phonebook
� Added support for displaying phonebook stored name instead of "From" header or "P-Asserted-ID"
� Added support for challenging Broadsoft remote-reboot NOTIFY (replies "401 Unauthorized" with WWW-Authenticate header).
� Fixed system ring tone does not play with accounts 2/3/4 when account 1 is set to use a ring tone that does not exist
� Fixed we do not use the same Authorization credential in ACK as in INVITE
� Fixed we do not display the number of messages (MWI) correctly when the number exceeds 99 (reported on Wiki). We now displays up to 999, messages over 999 will be displayed as 999.
� Changed behavior to "No Key Entry Timeout": imposed a minimum of 1 second and extended maximum to 30 seconds. Setting it to < 1 results to 1 and setting it > 30 will result in 30. Default is still 4 seconds. This is changed per frequent user requests (also reported on Wiki).
� Added under Broadsoft mode, register delay after 403 changed to 20 minutes
� Added support for reorder tone, played in lieu of busy tone when 403/480/484 is received for INVITE, Provisioning parameter P349
� Support G.726
� Fixed configuration download causes factory reset when EXT board is connected in certain scenario
� Changed behavior to "Automatic Upgrade": provisioning is delayed whenever a line is in-use, this include an offhook-idle line.
� Support BroadSoft Redudency.
� Fixed GXP-2000 HW1.1 cannot switch LED color in diagnostic mode
� Fixed iLBC will not work properly if switching from 20ms to 30ms without reboot
� Fixed 3WC with G.722 and G.711 bad audio
� Fixed digital volume gain management
� Fixed save volume audible by doing it in the background
� Added the function: In phone book->new entry, after input name and number, name and number can be displayed on �Add Phone Book Entry� menu
� Fixed the GUI MENU title incorrect in phonebook menu.
� Fixed name and number cannot be modified after an entry is added
� Fixed a potential DNS SRV priority handling bug
� Added under Broadsoft mode, keep-alive packets are sent to ALL DNS SRV resolved hosts
� Added DHCP option 61 (client identifier), Removed DHCP option 57 (maximum DHCP message size)
� Added support for Broadsoft Redundancy package, tested Broadsoft R14 test plan (113-123)
� Fixed BT-200 UI MENU option 7 (codec select) cannot choose G722/G726-32/iLBC
� Fixed HTTPd returns 404 when login as user (not admin) and saves any changes instead of displaying the page "Your configuration changes have been saved"
� Fixed BT-200 stops mute-indication when switching from speakerphone mode to handset mode and back to speakerphone mode
� Added "Hd" item to the BT-200 UI MENU option 8 (code rel) to indicate HW revision
� Fixed we do not follow the "a=fmtp:18 annexb=no" line in SDP (disable VAD), port from c64
� Fixed GXP-2000 phonebook GUI interface not initialized properly (empty title)
� Added support for G.722 (now interoperable with BT100)
� Added support for iLBC (interoperable with BT100)
� Added MENU UI codec selection support for G722/G726-32/iLBC
� Added support for customizable tones
� Provision parameter P343 (dial tone), P344 (MWI dial tone), P345 (ring tone), P346 (ringback tone), P347 (call-waiting tone), P348 (busy tone)
� Added default tone strings to WEB UI
� Added option to use remote contact in Refer-To solving the SPECIAL consultative transfer problem. Provision parameter P135/469/569/669, possible values 0/1, default 0.
� After factory reset, all accounts have codec selections in this order: PCMU/PCMA/G723.1/G729AB/G726-32/iLBC/G.722/GSM (consistent with other products)
� After factory reset, default NTP server is changed to ""
� Fixed DHCP sent DHCP discover from non-DHCP Client port (68) when network cable is disconnected and reconnected, introduced in with TCP/IP Stack, this is also a memory leak case which has small impact
� Fixed you can't hear the remote party during CW tones playing
� Fixed we tear down the call without sending BYE when we receive 488 for reINVITE
� Fixed we do not add CRLF for DTMF by SIP INFO in the body
� Relaxed the new TCP/IP stack about UDP connection check to allow incoming UDP from hosts other than the one we are connected to
� Audio updates
� Fixed overflow bug in digital volume scaling function
� Added 9 more dB of volume scaling gain range
� Reworked on VLAN handling
� Forced RFC2833 DTMF ending duration to be 100 if it is 0 (for compatibility with GXW when GSM is used)
� Re-architecture the audio component
� Fixed upgrade via TFTP through GAPSLITE fails
� Fixed BT-200 crash on incoming call by turning callhistory module's optimization off
� Added BT-200 support for mute indication by flashing the speaker icon
� Support for Event Notification Extension for Resource Lists (eventlist, RFC 4662) Provision parameter P134/P444/P544/P644 for accounts 1/2/3/4 respectively, allowing each account to configure 1 eventlist URI. You will have to configure each individual BLF monitored userid in the Basic Settings page and select the key mode as "eventlist BLF" (versus "Asterisk BLF") so no individual subscriptions will be sent.
� Fixed custom ring tones will not ring
� Fixed we used "dialog:id" instead of "dialoginfo:entity" in dialog XML to identify the dialog
� Behavior change: DHCP option 66 (P145) is default to 1 on factory reset
� Fixed GXP-2000 crashes on bootup after factory reset due to illegal freeing of some uninitialized localization strings.
� Fixed we updates Record-Route set by in dialog responses
� Fixed we do not follow Retry-After as indicated in 500/503 for REGISTER
� Fixed we respond to incoming non-INVITE requests with incorrect account when talking on a different account
� Added option to allow turn off display of in-call DTMF digits
� Provision parameter P338, default value 0 (digits displayed), possible value 0/1.
� Expanded syslog message to include DTMF type and ptime info in addition to payload type selected before session start
� Fixed TCP buffer-chaining bug which may be the cause of the HTTP download failure
� Support of DNS SRV query for TCP type (automatic--query for _sip._tcp.sip_proxy is sent when account is configured for TCP)
� Fixed we sent in-dialog requests to the proxy that we registered to instead of the actual proxy the dialog established
� Support Bellcore-drx (x=1-5) ring tones, supported as a general feature
� Support for BT-200 to disable call logs. Provision parameter P187 (for BT-200 only), possible values 0/1, default value 0 (calls logged as normal)
� Fixed config_e1/config_e2 page header not aligned
� Fixed GXP-2000 stalled when config_e2.htm is accessed
� Fixed GXP-2000 allows config_e1.htm access even when login using end-user password (123)
� Handset audio fixes
� Added bootloader version to HTTP header
� Reworked the SIP NOTIFY/events module to have a clean NOTIFY handler
� Added support for anonymous call rejection (per account); Provision parameter P129 (P446/546/646 for accounts 2-4 on GXP-2000), default is No
� Added support for Broadsoft remote-reboot via NOTIFY (check-sync event)
� Added support for remote reboot via NOTIFY for accounts 2/3/4 for GXP-2000
� Fixed we send 415 response for NOTIFY when no Content-Type header is present
� Fixed BT-200 fails to download configuration file (does not impact GXP-2000)
� Fixed GXP-2000 crashes when DHCP fails and link is down. Fixed when factory reset, Account 1 SIP Transport has no default value (should be UDP)
� Fixed the URI in auth header is trimmed for SUBSCRIBE
� Fixed GXP-2000 stay in speakermode after loopback call
� Fixed we will first send REGISTER to when DNS SRV is in use for SIP server
� Fixed a bug in SIP stack which caused some mysterious crashes
� Officially named the language pack file to gxp2000.lpf. As the name suggests, this file is model specific (gxv3000.lpf, bt200.lpf � etc. in the future).

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.2.23.pdf"

Bugs / Tweaks (

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

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

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

  • 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 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, always after about 1 hour - other providers are working fine...- Mirak
    • Note : (Dec30/06) For me, it happens with and only since firmware version 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 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 (, 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 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 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

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

  • 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"

(Oct11/06): Now moved to a stable release and available from the normal download page and directly here

Changelog (
  • Fix missing extension Module issue

Above changelog taken from the "Release_Note_GXP2000-BT200_1.1.1.14.pdf" file included with the firmware. This release appears to be a quick bugfix on, so other bugs may still exist.

Bugs / Tweaks (

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

  • MAJOR: (Dec15/06) No reconnect during ip renew from dynamic dns after 24h timeout: Since I updated the firmware to 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 it_IS_broadcast; For example, if your IP address is, netmask and gateway, a packet to 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 when was the latest stable and again when came out, but I have not received any reply (except for the "ticket generation") from them. The bug is STILL present in 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 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 Just tried 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 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 - 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 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 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 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 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 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 to this firmware. I have found that 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, when using g.711 codec. AGC in the prior ( 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

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

  • MAJOR (Sep26/06) Paging Issue In, Using asterisk , 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 - joekane

Firmware Downloads

NOTE: (Feb27/06) To upgrade directly to 1.0.2.x from any 1.0.1.x firmware you need the gxp2000.bin and boot55.bin files included in the and firmwares, just put the gxp2000.bin and boot55.bin files for the or firmware on the tftp/http server with the gxp2000a.bin and boot55a.bin files from the 1.0.2.x firmware. The gxp2000.bin and boot55.bin from the or firmware update the phone telling it to look for gxp2000a.bin and boot55a.bin ( they probally change other things too... ). - SoloFlyer

Created by: indesignfirm, Last modification: Thu 06 of Sep, 2007 (03:11 UTC)
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