To obtain the lastest firmware version of VoIP module, please go to links below. Older FW versions are available upon request. You can write to firstname.lastname@example.org
1. Save and unpack the zip file. FW upgrade has an extension *.update
2. In the web configuration tool of the IP door phone station, go to "Service", "Update Firmware" and via "Select" find the saved file on your computer. Then click on "Save".
3. The process may take up to 10min. Make sure you do not work on any other application during the process to avoid crashing of FW upgrade in the VoIP module. After the process has finished, please click on "Restart", which appears only after the end of the upgrade. The current firmware version numbers are displayed in the "Service" at the top right.
4. Please note not all FW update versions of the VoIP module are suitable for upgrades of the VoIP GSM gateways. Before you make any FW upgrade in the VoIP module of the VoIP GSM gateway, write us an email and double check that the latest FW version is suitable for the upgrade.
VoIP FW for IP audio / video doorphones only
- IPDP Slim
- IPDP Modular
- IPDP Modular Antivandal
- IPDP Fermax (IPDP Cityline, IPDP Skyline)
- IP Bell
(Please note IP BOLD uses different type of VoIP FW)
VoIP-GSM routers/gateways: the latest version of VoIP FW for VoIP GSM/UMTS gateways is available upon email request.
Attention!!! If you need VoIP firmware for IP BOLD doorphone, please go to this page.
If you need support with one of the IP doorphones listed above, please send us an enhaced log file for analysis, here is a guide how to do it.
After FW upgrade, restart the doorphone, then go to setting video - click on default values and then save changes.
V2.58, 14.10.2016 - a zipped file, unzip it before uploading to the doorphone
V2.57, 13.9.2016 - a zipped file, unzip it before uploading to the doorphone
V2.55, 4.6.2016 - a zipped file, unzip it before uploading to the doorphone
- DHCP fix that worked badly when the network address is different than 192.168.xy while customer has two segments separated by the router IP addresses
- another G722 codec - wider compatibility test.
- correcting, when in the special case generates more sections of audio
- Function of the registrar server in P2P mode (calls from clients who do not ride without registration (Apple & Co.))
- Confirmation of the calling party only allowed codecs (option strictly prohibit troubled G.722)
- Added option to restart the video if you fall out images from the camera
- to RTSP SDP for H264 added parameter 'FMTP' containing 'profile-level-id'
- repaired SIP Redirect (Moved)
- detecting In-band DTMF
- After reboot starts sending syslog messages, if required
- final version
- Unifies end the HTTP headers to CRLF (on wish customers)
- Check PBX Bypass (DNS, port)
- Prolonged ringing outgoing SIP calls (not the bottom doorphone!) For 5 minutes
- Replaces support in INVITE
- Update audio parameters When HOLD and Transfer
- Improved compatibility MJPG video with phone Innovaphone
- on the 'Video Settings' can be specified payload type for H264
- on the 'Video Settings' is a new field 'Compatibility' with a selection of 'Yealink' or 'Standard'.
- for compatatibility video with newer phones Snom added parameter 'state = "relevant"'
- on request 'INFO', which has unknown 'Content-Type', '405 answer doorphone Method Not Allowed '
- in P2P mode in the 'Network Settings' added the 'NAT address'. In it you can type 'public' address of the router which doorphone is hidden from the view of 'public' Internet. If routed ports for SIP and RTP in the doorphone goes in P2P mode and call out in a private network (if not routed, of course, can only call out).
- compatibility H264 video with telephone Yealink
- supported codec G.722
- typo 'Manual' -> 'Manually'
- added to http://video.jpg http header 'Content-Length'
- the parameters of the video cameras are percentages
- customization parameter
- The SIP BYE puts the correct authentication type by status code in the error message
- If still come after the BYE request for authentication establish appropriate answer
- After RE-INVITE updates telephony-event payload number
- Removed caching DNS SRV record sip server
If the SIP server is given ip address - nothing happens.
If the SIP server is defined by name:
- If it is entered on a port number (default state) DNS searches the guest directly (as yet)
- If the port number is not specified, the server name is considered a domain and that domain's DNS SRV record searches for locating SIP server
1. in the 'Switches' new field 'Enable external codes' where you can choose:
- Always enabled
- Illegal ever
- Allowed only when outage LAN
2. When receiving SIP Status '302 Moved Temporarily 'performs redirection on the' Contact 'from this message
3. for an incoming call sends a '180 Ringing 'or '183 Session Progress
- Move to new hardware
- FW 2.xx is applicable to all VoIP modules (backward compatibility)
- FW 1.xx is applicable only to older type of VoIP modules
* You can identify the type of your VoIP module on the image below
Changes on HW and SW
- SW echo limiter (settings on page Audio - HW suppressor can be switch off by turning left the trimmer ECHO)
- H264 compatibility with Cisco phones (CM-V8x)
- Width of the field in numbers memory increased to 30 characters
- Password under 'SIP settings' replaced by asterisks
- Video multicast (multicast address must be filled in the menu Video)
- DTMF payload adaptation Mitel phone devices
VoIP GSM gateway in the dial mode
- For one sim card gateway enables transmission of DTMF GSM-> VoIP during a call.
- At dual sim card the gateway is DTMF during a call is blocked just as before
1. events.txt remctrl and video can be put together with the password (page 'User Interface' field 'Video password protect'). Required in Popup.
2. added check OutboundProxy SipProxy
3. added event 'Registration' to events.txt (for PopUp)
4. recreating the file because events.txt troubled history records may be missing (for PopUp)
1. treated incoming overflow front the network
2. SDP parameter in the 'telephone-events' changed from 0-11 to 0-15
3. call transfer (REFER, Refer-To)
1. improved resynchronization after a jump in timestamp or loss mark a bit in the RTP audio stream
2.URL 'http://ip/cgi-bin/remctrl.sh?id=aloop' turns on the call that starts during the next 1 minute "loopback". Audio from the VoIP caller will be in guard / gate is routed to the caller's ear. To test the patency of the audio stream. (required UDVguard program).
3. UDVguard program can get information about the functionality of cameras, release SW in UDV module and registration status
1. when coming BUSY from GSM gateway not hang, but the caller will beep busy tone to the SIP ringing timeout expires
2. when you register (gate and doorphone) process SIP header with the date and time of the SIP server (set in the time to guard even without NTP time server). Respects the time zone from the Service menu. NTP has priority, if specified NTP server, the date of registration ingnoruje.
1. increase the reliability of storage of all Web setup
2. eliminate the possibility of rewriting the current settings by storing multiple (multiple windows, multiple users)
1. color adjustment settings for video and progression restart the CSS styles, so that they can customize
2. User interface modification in order to customize default values
3. Call log & Index logos are opened in separate windows
4. default timezone is GMT 1 (Central Europe)
5. repair: not switch day / night in guard by interval according
6. repair: When using DHCP during upgrade lost network connection
7. DoorPhone in the mode selection does not appear in the DTMF SIP settings, because it is unnecessary
8. improved selection / page (or video setup) based on the 'User Interface'
1. DTMF adjusted according to new specifications (customer)
2. Registration error in Austria (423 Interval Too Brief)
3. Items 'Sensors' included in the customization
for versions higher than V1.63 use for upgrade firmware module doorphone always the new download firmware for the module of the doorphone
1. the possibility of customization other items
2. another, better DHCP client
3. change codec 'None' on'-----', not need to translate
4. the menu 'Video settings' option added to the priorities of H263 and H264
5. 'User settings' sacked 'in Video call' replaced by section 4
1. correction, while active the video call is on some phones sided audibility
1. patch registration 2. patch outbond proxy
3. customization to the next Field
4. repair video on the front page
5. fix the video display fields
6. 3 attempts to finish reading out data from the doorphone, before the reports an error
7. to the extended log stores information about the time of image processing
1. H.264 is possible use if set in User interface
1. automatic renewal of connecting a camera (video solves failures of the interference)
2. 'User Interface' 'Push Video' for SNOM phones (Snom870 etc. ..)
3. Default camera parameters changed for a new camera model Microsoft
1. Remote control relay (remctrl.sh) given the password
2. removed from the gate:
- a redirection module
- start module
- modules directly
3. released guidelines has now been addressed in the VoIP software in accordance with the table LCR
4. under 'Network Settings' added option 'select mode'
4.1 single-gate (one of a VoIP gateway)
4.2 Two-channel gate (similar to 'modules directly')
4.3 Dual Gate (one of two GSM VoIP channels)
5. at one SIM gates goes under 4.1 and 'selection mode' is not offered
1. - DTMF for Oxo
2. - 'User Interface' election 'Video on the password' secure / video.jpg and / video.mjpg password
3. - bugfix: video introduction always require a password
4. - the choice of P2P means star in telephone number = dot, the choice of the SIP proxy server means star in telephone number = star
1. SIP server without prefix (for single-channel gate, or automatic dialing channel is an optional prefix. Then if the number called does not the local user, is directed to GSM)
2. SIP NOTIFY (Red Cross)
3. rfc3263 (sipserver by domain) extension of the IP in network settings
1. reduced level of at least four buttons on a button (these options and properties are given by Customization)
2. Added authid (UserID)
3. "Network Settings" | "Hostname" to the SIP Display Name
4. added codec 'None'
5. fixed sequence of codecs
1. when upgrading (from this and later versions) will be displayed per cent, but the message informing
Only for GSM gateway:
2. added item 'Choice module' which can be filled so as to function as LCR. Complete the initial part of the number (or interval) and the operator of the modules for the operator to use and whether to allow the overflow when the module is already assigned to another module. It's basically pretty simple.
If the function is turned on, but leave the table empty, the module will be one and the overflow is allowed.
If the table is filled and no line called number fails, the call will fail.
- While (if an automatic selection module) in the menu 'SIP Server' appears only one field for a prefix for calls to the GSM common to both modules (the correct module is selected automatically)
3. Brief help for 'Select module' has been added to 'Help'
1. improved interaction with the pop-up video
2. added NTP time zones, Australia, New Zealand
3. remove the error message 'mount ...' the firmware upgrade
1. Function with 'realm' is equal '' (OXE)
2. He have best reaction on RTP Mark
1. On the 'SIP settings' is a separate field from the registration server box (outbound) proxy server. (Where the registration server stays empty, used as the default proxy server)
2. rules for bidirectional calls while logged on OrangeSK.
1. Allows you to enter a code to switch the day / night, and for switching the lock for hanging one, for example, closing switch 1 is 5 - set codes *5.
1digit code examples - you dial for activation 4 - set code is *4, you dial for activation 0 - set code is *0
2digit code examples - you dial for activation 45 - set code is 45, you dial for activation 00 - set code is 00
2. Switches for switching the button on the toolbar can be entered into the memory of buttons *** 1 for the first switch or *** 2 for the second switch your phone number or location. IP addresses
1. OutBand DTMF, send over DTMF messages between SIP client and GSM modulus namely by either like SIP INFO, or in RTP (it is possible choose over web, page ' setting SIP ' ). In direction GSM- >SIP only in canal ' A' at ' DISA'=0
2. field MSN can contain also letters, changed type from int on char, max. longitude 4 characters, case sensitive. Functions in P2P also in sip server.
3. clarify interval registration. Value engaged over web sends to the register server, send REGISTER request them of 10s shorter (to on server didn't happen to unregistration).
4. GSM parameters T4 and T5 are emptied.
5. possibility set headline web window over customizacion. For example is attachment of customizacion, which will set headline and dhcp_id on ' DoorPhone
Warning!! - for use old file with style you need add to style.html this text.
1. video for Snom
2. expansion style, so so as to be possible to record TCS like customization without changes of firmware
3. added label for Help ' daily interval' and ' UI', writed delivered en mutation and translated into Czech
4. changed way of updating time to ' daily interval', well - tried in Moz.(Lin and Win) and IE6
5. button ' Extended log' has ' intuitive' title, appearance to frequent obscurities
6. added possibility switch off lighting to the ' basic setting', (PROG41x) - firmware in DoorPhone 5.9 or 2.9 and higher
7. removed dependence on NTP in interval registration (NTP scatter time in modulus)
8. to the SIP headers added parameter ' Allow'
9. corrected error, when while using DHCP stop away call (bypass gw ' ' check failed)
10. quicker rtp synchronization at coming RTP.Mark (Lithuania)
1. solved control of relay in RFC2833 for Alcatel OXO
1. solved switching on exchange Astra (with notification and without). In ' setting SIP' be necessary switch on ' symmetrical RTP' (default off). As well at switching on exchange Cisco switch on ' symmetrical RTP ' .
1. elimination drop - out talk at some types switching
2. removed deafness at connection between by both halfs of gate
3. bettered speed of response (how SIP, so RTP)
4. on page ' ;Setting SIP' ;
- choice ' ;180 Ringing' ; or ' ;183 Session progression' ; some clients want to it and another want it
- choice ' ;Symmetrical RTP' ; it needed Cisco and other it more likely effects problems
1. version send temporary not available if call not possible make via GSM gate
1. ntp supports 'perhaps' all ntp version
3. uncontroll 'rtp data timeout' before pick up (TWIN problem)
1. repair error RTP - video on CM Cisco
2. week atutomatic switch Day/Night mode
3. voip monitor
4. save and restore settings parameters
5. disable / enable video in RTP stream in call
6. disable / enable video on titul site on web browser and safe by password
7. possibility change port of web sites
8. possibility disable connection by telnet
9. prolonger password (max 100 digits)
1.repair error RTP - MediaType - automatic detection (OXO)
V1.31, 14.9.2009 stable derivate from version 1.27
1. send 183 Session progress in place of 180 Ringing at preanswer
2. on reply 401 Unauthorized no-respond hang-up, but as a on 407 Auth. required, proceed authorization and continue in calling
3. unused for registration port reserved in web sites
4. it's no go recording upgrade firmware
5. no sending right CLIP
6. to the www append' Roaming:
1. innovation beep before dtmf
2. ringing after task dtmf
3. empty dtmf -> default extension
4. remove freeze at reading sites' sip server' and' user'
6. odstraneno field' rec' in' gsm gate' (grafity p. )
7. append field' roaming' in' gsm modulus 1 and 2'
8. dtmf it is possible early (before timeout) actually press' #'
9. integration tolerance full path to recording style
10.first REGISTER sending on engaged port
- improvement picture from cameras brondi
- CLIP for IPEX at call over external SIP server
- version with innovation continuousness and time-lag of video.Tested with cameras Brondi and Microsoft
- return original driver for camera brondi - picture in substance almost not break down
- Correction sending CLIP at using external SIP server -check functionality cooperate with Popup sw in mode P2P and with SIP server
- in www append verification current data using int and ext SIP server
- enlargement detection USB cameras in doorphone
- added pause, until finish start of camera
- always it is impossible perform upgrade firmware on lower version