ST2030 SIP Firmware Versions

Thomson ST2030
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ST2030 SIP Firmware Releases

Current firmware releases for SIP available at: http://www.technicolorbroadbandpartner.com/telephony-solutions/products/product-detail.php?id=87


SIP Firmware 2.74 (21/02/2011)

Downloadable at http://www.technicolorbroadbandpartner.com/getfile.php?id=7373

SIP Firmware version x.67 (x=1 for normal, x=2 for new "secure" version)

Downloadable at www.thomsontelecompartner.com. Direct Link (might not work if out-dated)

Known Issues

  • Phonebook backup file doesn't carry any ringer information


SIP Firmware v1.66

http://www.thomsonbroadbandpartner.com/getfile.php?id=6466

Known Issues

  • 21/01/2009 - linuxmaniac - Autoprovision config for extension keys does not work. (FeatureKeyExtXX=)


SIP Firmware v1.58

http://www.thomsontelecompartner.com/getfile.php?id=4718

Known Issues

  • 30/01/2008 - gmaoret - After an upgrade from 1.54 to 1.58 the "outcalling" volume is too low. There is a way to change the microphone gain/volume?
    • 21/02/2008 - stoffel - volume in/out is fine here (hw rev. 0)
    • 21/03/08 - Check that you upgraded with both v2030SG.071214.1.58.6.zz and v2030_dsp_v101.zz before the TelConf. I've had low "outcalling" volume on earlier firmwares too; that was when that "dsp.zz" was not installed.
  • 30/01/2008 - voermans - Because of the changes in de TelConf-file, you first need to upgrade to 1.57 (including the correct TelConf), before upgrading to 1.58! Also make sure to always upgrade the TelConf file also!
  • 06/02/2008 - nebz - Phone refuse call with hidden caller-id... have to downgrade to 1.56 (maybe 1.57... not tested)
    • 21/02/2008 - stoffel - my v1.58 works with unknown caller id's
  • 21/02/2008 - stoffel - Transferring a call back to a phone does not work. Example: external line gets picked up by phone 1. Phone 1 transfers to phone 2 with an attended transfer. Phone 2 then talks to the external party and tries to transfer back to phone 1 with an attended transfer. There is no TRANSFER button on phone 2 if that phone enters the extension of phone 1.


SIP Firmware v1.57



SIP Firmware v1.56



SIP Firmware v1.54




New Features

  • Provisioning via https is now available
  • Automatic Hang up: allows hanging up automatically ‘x’ seconds after the call is remotely ended. See companion document for details
  • Dutch language added. See Configuration File Syntax document.
  • Portuguese and Slovenian tones supported. See Configuration File Syntax document.
  • The parameter “AutoTurnOffSPKTimer” has been deleted. This parameter must be suppressed in the config files. If not, errors will happen during APS.
  • CallParkType parameter: it can be 0 or 1. It is used to differentiate the two modes of implementation for call park/park retrieve feature. Default is 0 (draft-sipping-service-examples implementation). See companion document for more detail
  • APS upgrade between SIP and MGCP is possible if boot code 1.11 is used, and from/to versions equal to or beyond: SG1.54, MC1.53, MX1.52. See companion document for details.
  • It is now possible to avoid APS trying to download the firmare. See companion document for details
  • Ringer can now be heard on headset as well. See companion document for details
  • Ringer can be turned off now. See companion document for details

Improvements Made

  • Call back (CCBS): although the feature was working, Call flow included unnecessary signaling which has been removed.
  • Remote phonebook: intermediate url’s with more than 115 chars will not cause misbehaviour now.
  • ST2030 stores properly 2nd, 3rd, etc incoming calls in the call log if there is an active call. Dialed or missed calls are correctly logged.
  • BLF-Error responses to a NOTIFY does not delete the subscription on notifier side.
  • From headers like sip:1234;pn=34@domain.com will be correctly parsed.
  • Contact headers like sip:1234;pn=34@domain.com or sip:+1234;pn=34@domain.com will now be correctly parsed
  • UTF-8 now used for name encoding, which will no longer cause wrong display on the other side when special chars are used.
  • Transfer: ST2030 as a transferer will no longer send a wrong Replaces header if in a previous transfer as a transferor it did not receive BYE from the target.
  • BLF will now work if xml body contains tab
  • BLF in conjunction with keepalive via OPTIONS is ok now
  • It is now possible to perform a blind transfer even if target’s number matches the dial plan
  • DTMF configuation is accessible now via common/MAC files, instead of Telconf only.
  • Several wording errors in Norwegian language corrected

Known Issues

  • Call Pick-up in Asterisk environments: in order to perform Call Pickup in Asterisk environments-within or outside BLF, a patch for Asterisk server is needed so that Dialog information is provided in accordance with ST2030 implementation.
  • From headers like sip:+1234;pn=34@domain.com will not be correctly parsed, call will be shown as Anonymous.
  • The display doesn't show the name of the incoming call when received display name starts with the letter "O" (uppercase).
  • APS: option 150 must be now binary encoded. String encoding is not handled. Will not be changed.
  • DNS-Srv: port information in DNS-srv answer not taken into account.
  • Long Refer-to headers with user information above 63 bytes will cause the transfer to fail.
  • Re-INVITEs from an anonymous caller do not include a Contact header.
  • APS-Telconf not automatically downloaded when reset to default procedure is executed.
  • Volume is not saved after reboot.
  • NTP Server with FQDN, the clock doesn't progress after reboot. Workaround: use an ip address to configure your NTP server
  • ST2030 does not support SDP with m line with "text".
  • When editing the name of the proxy in the telephone, if it is longer than the display it doesn't allow erasing it completely. The same with predial > 24
  • In call-back scenario, ST2030-caller removes the ringing tone and opens the audio path (r1.53 keeps the ringing tone until picks-up the call too). MMI still shows the CCBB msg.
  • Call park server does not allow special characters like “*”.
  • If you use protocol DNS with SRV records some misbehavior will probably occur: after a reboot, phone will not get registered for a while unless we press "Apply" in profile menu via web. The phone in some cases is not performing DNS query (type A) for this highest priority record but it does it properly for the rest of the records in the SRV response. Workaround: introduce in the DNS+SRV server a duplicate entry for the location with the highest priority (priority=1), or provide several valid locations.
  • Remote address book doesn't work, the phone doesn't make the http request to the specified server. Thi for firmware upgrade from version 1.53


SIP Firmware v1.53


New Features

  • Talk and hold event packages for click-to-answer and other 3PCC scenarios.
  • Call-info “answer-after” parameter for intercom and click-to-call scenarios.
  • Phonebook: it is now configurable to use or ignore domain name when checking if caller/called party is in the Phonebook.
  • Refer-to header in Attended Transfer: it is now configurable whether to use Contact of the 2nd dialog or request URI to populate Refer-to header.
  • Network conference for Tekelec T6000 softswitch.
  • APS enhancements to avoid reboots when no update pending.
  • VLAN configuration via DHCP improved with a persistency flag.

Improvements Made

  • If a call is done in predialing mode, dialed number will now be stored in Redial list.
  • Attended transfer working now when transferee is located behind Alcatel OXE gateway (dialog 2 contact is an ip)
  • ST2030 Transferee: transfer will no longer fail if Transfer target has CF service enabled.
  • Transferee in local Anonymous mode will no longer reject REFER, so transfer will take place.
  • REGISTER: 3xx responses processed in this release.
  • SIP over TCP fixed
  • Fragmented TCP now supported
  • DTMF RFC2833 when 18x with sdp received including OOB information: error cases corrected.
  • Line LED no longer goes off during transfer procedure
  • ST2030 no longer cancels a call sent to another phone which belongs to the same calling group (Asterisk)
  • SDP containing audio and video is now properly handled
  • SDP contained in a reliable provisional response is now considered a valid sdp answer
  • DNS srv answers containing multiple addresses are now properly handled
  • OPTIONs requests within a dialog are now accepted
  • APS-more than one ip accepted now in option 150 (first one will be used)

Known Issues

  • Call Pick-up in Asterisk environments: in order to perform Call Pickup in Asterisk environments-within or outside BLF, a patch for Asterisk server is needed so that Dialog information is provided in accordance with ST2030 implementation . This patch is under preparation and will be released soon.
  • DTMF setting via Common or MAC files is not possible any more. It has to be done via the Telconf file, with a specific syntax.
  • Please do not use empty parameters in provisioning *.inf file. All parameters are mandatory except melody, system melody and Call waiting files. If you do not wish to load them, just remove the parameter from the file
  • Call back (CCBS): although the feature is working, Call flow includes unnecessary signaling which will be removed
  • DNS-Srv: port information in DNS-srv answer not taken into account
  • APS: option 150 must be now binary encoded. String encoding is not handled
  • Transfer: ST2030 as a transferor will send a wrong Replaces header if in a previous transfer as a transferor it did not receive BYE from the target
  • Transfer-Blind transfer with dial plan matching numbers is not possible
  • Long Refer-to headers with user information above 63 bytes will cause the transfer to fail
  • Remote phonebook: intermediate url’s with more than 115 chars will cause misbehaviour
  • UTF-8 not used for name encoding, which will cause wrong display on the other side when special chars are used.
  • From headers like sip:1234;pn=34@domain.com will not be correctly parsed.
  • ST2030 does not store 2nd, 3rd, etc incoming calls in the call log if there is an active call. Dialled or missed calls are correctly logged
  • Re-INVITEs from an anonymous caller do not include a Contact header
  • APS-Telconf not automatically downloaded when reset to default procedure is executed.
  • APS-randomly, in APS’s where only Telconf is loaded, phone will not properly reboot. Manual reboot needed.
  • BLF-Error responses to a NOTIFY will delete the subscription on notifier side


SIP Firmware v1.52.1

New Features

  • DHCP Option 77 (User Class Identifier) is now supported. See additional info.
  • List-URI oriented BLF is now supported. See additional info.
  • Administrator can now disable services such as Transfer to Vmail and Call Pickup, on top of previous set of configurable features. See additional info.
  • MWI: generic message displayed when number of messages is not provided by Voicemail server
  • Domain name is now hidden in Call logs as well when the feature “Hide Domain Name” is enabled
  • Phone number shown in idle screen can be disabled via configuration. See additional info.
  • Call progress indication when the remote is in alerting state can be disabled via configuration to avoid confusing info in cases where Busy indication is done via networky announcement. See additional info
  • Blind Transfer: enhanced early BYE support
  • Refer-to header in Attended Transfer: Contact of the 2nd dialog instead of the request URI is now used to populate the Refer-to header.
  • DNS redundancy : registrar fall back time reduced in order to allow authentication registrations to succeed before register transaction timeout has expired

Improvements Made

  • Speed dial key values will no longer be lost if you reboot the phone
  • The following issue no longer happens: "Wrong Number!" in predialling when VoIP dial plan was used: if the dialled number in predialling mode matches the VoIP dial plan, the phone showed the message "Wrong Number!" instead of "Connecting...".
  • Spurious DNS queries: in various SIP service configurations, spurious DNS queries are no longer launched before Cancel and Subscribe requests.
  • Wrong DNS flag corrected. Flag-z in DNS requests was enabled, which caused problems with some DNS servers.
  • Call Waiting - 2 calls with the same From username are accepted now-including 2 anonymous calls.
  • UK tones corrected
  • DND and Lock softkeys can be disabled now
  • Shortcuts to disabled options are not possible now
  • user=phone is now included in request uri's. If you need to disable this parameter, please use USRPhoneFlg=0 in sip
  • SIP NOTIFY-Check sync mechanism recovered in some failure scenarios
  • Reset to default now lauches complete APS from either 2-key sequence, mmi or web gui
  • RFC2833 digits sent when in handsfree mode, no matter what the value of “tone_out_power” in Telconf
  • Dialing “#” as normal digit now will include the escape sequence “%23” in Request –URI and To user uri
  • http chunked responses now supported (xml phonebook)
  • NTP: time no longer incorrect between noon and 1pm. If 12h format is enabled, then between noon and 1pm the phone will now show the time as 12:xxpm
  • DHCP no longer randomly not using VLAN settings at reboot (DHCP DISCOVER).
  • NTP: MMI no longer shows incorrect time in case of long term network problem.
  • MMI interface for VLAN settings --> "Pri-S" which had no use has been removed.
  • Registration: If header "Event: registration" is present in a 200 OK to a REGISTER, the phone n olonger ignores the 200OK and resends Register request. Registration now succeeds in this circumstance
  • Transferee no longer rejects the REFER msg if -Refer-to contains Tags in specific order
  • MWI led status is now correctly updated and maintained
  • CCBS with targets not supporting this feature is now working again.
  • Redial with message on the display (like Missed Calls or Message Waiting) is now possible.
  • Subscribe to MWI: refresh in case of long term network failure has been corrected
  • Eleventh call will no longer receive a “404 Not found”, but a “486 Busy”
  • Bad codec negotiation observed with UPDATE which had been observed when compressed codecs were involved is now correct.
  • CANCEL no longer includes Route headers
  • Blocking corrected for some conference Asterisk situations
  • Hide domain name feature when the From header is like From: number <sip:1234>;tag=xxxx is now working properly

Known Issues

  • Call Pick-up in Asterisk environments: in order to perform Call Pickup in Asterisk environments-within or outside BLF, a patch for Asterisk server is needed so that Dialog information is provided in accordance with ST2030 implementation . This patch is under preparation and will be released soon.
  • DTMF setting via Common or MAC files is not possible any more. It has to be done via the Telconf file, with a specific syntax (refer to documentation)
  • Please do not use empty parameters in provisioning *.inf file. All parameters are mandatory except melody, system melody and Call waiting files. If you do not wish to load them, just remove the parameter from the file
  • If a call is done in predialing mode, dialled number will not be stored in Redial list. A fix will be available before next release.
  • Attended transfer does not work when the transferee is located behind Alcatel OXE gateway. A fix will be available before next release
  • ST2030 Transferee: transfer will fail if Transfer target has CF service enabled. A fix will be available before next release.
  • Call back (CCBS): although the feature is working, Call flow includes unnecessary signaling which will be removed in next release
  • Transferee in local Anonymous mode will reject REFER, so no transfer will take place.
  • REGISTER: 3xx responses not processed in this release.
  • Line LED goes off during transfer procedure
  • SIP over TCP has limited performance in this release
  • DNS-Srv: port information in DNS-srv answer not taken into account
  • DTMF RFC2833 when 18x with sdp received including OOB information: some error cases detected.


SIP Firmware v1.50t3

New Features

  • Auto Call (Emergency call): when enabled, ST2030 will immediately dial preprogrammed number when you go off hook.
  • Missed Call indication enable/disable: this will allow administrators to avoid the phone showing the number of missed calls. For environments with shared lines.
  • Autoanswer feature: headset or speaker configurable
  • Long press with Menu key is now a shortcut to MenuàUseràInformation. For Support convenience.
  • Manual Daylight Saving value for the corresponding flag. On top of previous values off/auto, this new value allows manual DST setting for administrators in areas with non-standard changing dates for DST
  • NTP value can be derived from DHCP or saved value configurably
  • Portuguese language support added

Improvements Made

  • Call hold from callee side: standard behaviour restablished.
  • Various BLF limitations listed for SEG1.47 which have been fixed and are no longer an issue
  • Configuration: “Outbound proxy with pre-existing route” is again functional in this release.
  • Codec negotiation problems with G711 fixed (Erroneous 415 Unsupported Media or 404 Not found messages)
  • Double authentication and all-request-authentication scenarios have been improved
  • HTTP requests (provisioning and remote phonebook) port handling implemented
  • Hold: Music server/Local hold selection now operative. When Music server is selected, received RTP stream will be played.
  • RTCP now sent in hold state also
  • ETSI SIP conformance cases SIP_CC_OE_CE_TI_005, SIP_CC_TE_CE_V_008 SIP_CC_TE_CE_V_010 SIP_CC_TE_CE_V_011
  • Call transfer when transferor receives 180/183 with SDP is now working
  • Call state indication when 180/183 with SDP is received (normally external calls with gateways involved) is now ok
  • DTMF out of band in Handsfree mode not sent: fixed, make sure to use the Telconf included in this release
  • Reset to default will now force Common file to be downloaded (if APS enabled).

Known Issues

  • Speed dial key values will be lost if you reboot the phone-if they were configured using the keypad and a domain is not provided. It will work fine if configured using the Web GUI and/or domain is provided.
    • Workaround: always include the domain name, even if it is the same as yours, or use the web gui
  • Call Pick-up in Asterisk environments: in order to perform Call Pickup in Asterisk environments-within or outside BLF, a patch for Asterisk server is needed so that Dialog information is provided in accordance with ST2030 implementation.
  • STUN will not work for Restricted Cone and Port Restricted Cone NAT types. It will work fine in Full Cone NAT environments.
  • Reset to default using web gui or phone menus will not trigger complete APS (all files). However, Reset to Default using shortcut (Press headset+Mute key while power up) will launch a complete APS.
    • Workaround: use the 2-button-powerup shortcut
  • "Wrong Number!" in predialling when VoIP dial plan is used: if the dialled number in predialling mode matches the VoIP dial plan, the phone shows the message "Wrong Number!" instead of "Connecting...". The call goes through after that.
  • Spurious DNS queries: in ceratin SIP service configurations spurious DNS queries are launched before Cancel and Subscribe requests. If there is not a DNS server answering (an error response is fine), this will cause some delay.
    • configure an outbound proxy equal to your proxy
  • NTP: time incorrect between noon and 1pm. If 12h format is enabled, then between noon and 1pm the phone will show the time as 12:xxam
    • Workaround: enable 24h time format.
  • Call Back incompatibility with Asterisk (due to id param). When ST2030 sends SUBSCRIBEs for CCBS, it is adding a id param inside the Event header (Event: dialog;id=1) which is correct, but Asterisk does not recognize it and rejects the Subscribe with 489 Bad Event.
  • XML directory: chunked http responses not supported.

Created by: stgnet, Last modification: Mon 18 of Jul, 2011 (14:09 UTC) by laughlen


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