There are firmware for the VoIP module.
The downloaded firmware file you save in your computer.
For easy download in *. ZIP archive, so you save the contents of the archive and unpacked *. zip.
Firmware upgrade for use in IPDP, always has extension *. update
Downloaded file with firmware save in your computer for easy download is in archive *.ZIP, so that save the file as far as unpacking subject archive *.ZIP. Firmware upgrade by force of WEB site interface in Blue Gate SIP (WEB browser in your computer - write IP address of Blue Gate SIP - clicks on "Setup" - name admin password 1234). Step by step manual in this file.
The current firmware version numbers are displayed in the "Service" at the top right.
Clean imported language files: use this file - lang_delete_all.cust
Set to default language (Czech and English): use this file - lang_set_default.cust
- apply this files in ZIP like as firmware
Basic file for translate English to another language with help. Translate only expression in quote and preservation mark of HTML formatting.
- use this language file En
Now is possible use FW V2.17 and higher for new hardware of IP module
- Move to new hardware
- FW 2.xx is applicable to all IPmodules (backward compatibility)
- FW 1.xx is only applicable to older IPmodules
* What is your IPmodule you know from the picture
Changes on HW and SW
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. 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. DTMF adjusted according to new specifications (customer)
2. Registration error in Austria (423 Interval Too Brief)
3. Items 'Sensors' included in the customization
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. 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 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)
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
- h263 video, 10pict./s.
- with camera Brondi picture cut or fill black frame, according to setting of picture parameters
- old camera can't smaller picture, so that always display cut out (zoom)
- to the LOG registration is append relevant reason failure
- to the LOG call is append reason rejection incomming call
- correction error with calling GSM->Internal SIP server, which express as only at running with once GSM gateway
- append receive DTMF in SIP message INFO to the existing receive DTMF in RTP
- change parameters S7
- after default setting is internal SIP server off
- finish "Help" in czech and english
- loudness sound reducing on halves (about 6dB)
-' bypass SIP server' error, resolved DNS .)
- in new version is correction error, when was talk rejection with thus, that' Bypassing SIP server' therefore are have had address SIP server engaged by the name (far from it IP address). Now is append DNS resolver and ought it function with name and IP address.
- append check entering address in menu setting of network, address gate1, gate2 and internal_sip_tear down as far as are dated up - must be every another
- check setting sip server, prefix must be engaged if is engaged corresponding ip address in menu setting_of network and every prefix must be another
- betterment check circumvention sip server
- correction savings parameters' control at incomming call' in doorphone
-repair registration and call to the innovaphone
-in P2P mode is possible paste before IP adress number of extension