ST2030 feature wishlist
For features that were on this list and are now implemented, see ST2030 FAQ or ST2030 SIP Features
New Features Wanted
- Implement Called Party Identification
- When from a telephone make a call to other, in the phone that make the call show the name of the called phone like the patch( http://bugs.digium.com/view.php?id=8824)
- Implement Sip registrar fall back
- The phone with the primary and secondary sip registrar configured, if the primary is down the phone register with the secondary, but when de primary is up again the phone not reregister with the primary itself, except if you manually reboot it.
- Intercom feature: auto answer via sip header (if header contains 'intercom=true')
- Auto answer on second call if not using speakerphone for first call
- Automatically hangup phone when call ends, rather then play fast busy tone until user takes action to hang up call
- For speakerphone, hang up automatically, possibly with alert tone
- For headset, hang up automatically, possibly with alert tone
- For handset, no change needed
- Macros in config, so that we could for example send a generic dhcp option to phones to request an URL, and the the phone would fill in it's MAC address in the request.
- Thomson: We are working on PHP provisioning that should do what you are asking for.
- A backlight for the screen - for next generation... - Fabrice
- Thomson: Will be done in next generation phones. Too big investment to change this in ST2030.
- Although the cost of the backlit LCD is not great, it is physically thicker, and would require a tooling change on plastic injection machine, which is very expensive.
- DOCUMENTATION that covers configuration of more than the basics. Really basic stuff is obvious so needs little comment. The admin guide is sort of OK apart from the bad english and content mostly communicating the obvious. Lots of wasted space giving us scrrenshots of the Web UI with no commentary.
- The admin guide will be converted to a format on this wiki soon.
- More accounts active at the same time
- Thomson: Complicated to do properly. We will not do it as it better handled in the PBX.
- Option to disable display of missed calls, or ignore logging calls from certain numbers
- Thomson: Underway
- Better 'transfer support'. When having one active call, and other call comes in, you have to first answer the second call before you can transfer your first active call. The only way to get back control of the original call is to answer or reject the new incoming call.
- since firmware 1.60
- When on an active call, first call, if a second call comes in and you want to do something with the first call, options like "transfer","hold" and so on are hidden... it's true... but if you press on the "C" button (back button) while the second call is overriding your screen, the second call will stop blinking on the screen but will still make your phone ringing... you will just have one more function key blinking fast, meaning that a second line is ringing and in the meanwhile, your options for the first call reappeared on the screen. So you can by example now transfer your first call and then answer your second line when you're available... (suggested by Hadrien)
- Pre 1.60 work around: reject incoming calls and tweak dialplan to park rejected calls
- since firmware 1.60
- Browsing missed call list from an external server - Just like it is possible to use an external server for directory services : when an incoming call is forwarded to several phones, each keeps this call in memory so someone can be recalled several times. Centralizing this missed calls directory ease customization : when a call in answered once, it would disappear from other missed calls lists.
- Add syslog logging
- Syslog capability is not planned
- Tool to save phone configuration files in readable format. Currently configuration is saved in .zz files which cannot be edited nor read with standard text editing tools (notepad and so on). It would be very useful to be either able to natively save config files in text files or convert existing .zz files.
- Improve user settings savings - Just like Snom hardphones, it would be great if it were possible to allow at the same time, centralized provisionning and to keep somme user settings (ringing tones, ...) from being erased after phone reboot.
- Secure FTP - Anything securing FTP exchange, such as FTPS, would be welcome.
- XML Support - It could be very useful to design custom XML menus served by a web server and activate those menus at any time (on call, idle). This way, integration with Asterisk could be pushed a step further with functions like Followme, Malicious Call Identification, convenient Callback-on busy feature, etc.
- 802.1X - When physical access to network cannot be prevented (Universities, large building,..), including a 802.1X supplicant triggered during boot time would keep network from unauthorized access.
- Redesign BLF (Fx) keys to ease key labelling - it is difficult to insert a paper sheet holding BLF keys labels : the keypad shape is tortuous. Straight lines designs ease paper inserting.
- Option to call from the computer using a url like "http://ip_of_the_phone/?nr=012452351
- Thomson: Underway
- Allow to add other variables to the address book url for example Regid and regpwd to authenticate the user.
- Customize a function key to fetch a simple txt/xml web page and show it on the display, even better a scheduled function that check for the page if it changhes the fx key blink, when pressed the phone shows the page on the diplay.
- Allow to specify a default address book, other than local
- Avoid Prefix for received and missed call to be added when recomposing using the Redial key
- Allow to customize a Soft Keys for specific functions for example a telnet command to change a phone setting.
- Activate Ringer Off function from a Soft Key
- Blind transfer via line buttons. Unattended transfer should work with line buttons -> e.g press transfer(softkey) followed by line button
- Import-export local address book
- Anyone: I think this feature is now implemented
- Extend product range with higher end model : large, color screen, XML, gigabit (?), wideband codec, ...
- Open source firmware.
- SNMP support (done as of 1.62)
- SRTP-SIPS support
- Auto-answer number list or auto-answer on special SIP header.
- I notice many people want this feature for different reasons.
- Makes is possible to do more seamless originates from Asterisk.
- STUN issue: To solve many registering problems, please add these options:
- On boot, if stun is enabled, wait for stun response before register. Now, when we power up our network, the phone complete the boot earlier than the router and obviously get an error from stun. As soon as WAN is up, the phone registers with LAN ip.
- Repeat stun before every SIP registration. Sometimes ADSL WAN ip changes but the phone still register with the cached ip got from first stun response.
- In SIP setup, add an option to specify the local IP or DOMAIN
As-yet-unorganized entries from past:
- auto answer control via SIP Header (e.g. Call-Info: answer-after=0) in order to selectively enable the autoanswer feature by controlling them with asterisk dialplan
- I second this, this would be very helpful.
- implement broadsoft enhanced sip cti features
we are going to use the thomson phone in a call centre with a genesys/asterisk implementation and require
sip notify - talk and hold message. This will allow the phone to be controlled from a soft phone, sip answer
(in the headset) and put line on hold from the soft phone.
- what boardsoft features are currently available in 1.50 ? Is the auto answer and click to answer feature integrated yet?
- The auto provisioning work wonderfully, however,
user menu cannot be disabled either. It to be able to remove these
menus from auto provisioning.
- make the call waiting tone optional when using this feature, as the phone also lights up the keys and show a
Thomson: None wants the same thing about that... we will make it configurable
Ales Farcnik: At least two of our clients demand this feature. Others are annoyed by the beep. If you don't fix it soon we will be forced to change the phones in this companies, and we don't want to do this. Could you implement a quick fix that can be applied with APS, before you implement it in the web interface. Thank you.
- turn off missed calls log is also a common request by mine customers, possibly by a SIP-header? the same
Thomson: A parameter will allow that
- Now that the BLF indication works, it would be nice to be able to pick up ringing extensions. This can be
add a prefix to the dialed number if the key is in the ringing state (Snom uses *8, which would be just fine) -hehol
Thomson: Pickup will be performed just by pressing the key linked to the monitored extension
hehol: That's true from the phone's perspective. But the server you are sending the call to has to be
able to distinguish if you are trying to call the extension or pick up the extension. Since there is no suitable
SIP header or body, pickup has to be implemented through an extension prefix.
picciuX: i second hehol: and the possibility to specify the "pickup prefix" in configuration would be great!.
AppTech This is a real killer/painful problem in customer sites. hehol is very correct and endorsed.
Thomson: If the latest implementation of pickup do not suit your need, please give more detail on the call flow.
Fips: How does the latest implementation work? Nothing found in FW (1.47 and 1.48beta) docs.
sip debug just shows, that the phone is trying to call the extensions
picciuX: as already said: the best implementation of pickup, IMHO, should use a config parameter
to set a "pickup prefix" (it could be '*8', '**', or whatever else), that the phone will use to call a ringing
extension. Ex: extension 300 is ringing, and is one of my ST2030 BLFs. I see the button blinking, I press it,
and the phone dials <pickupprefix>300. This way the pbx can understand that this call is a pickup attempt
for 300, and not another call to 300 (which would be signalled busy).
Quintana: We have create patch for pickup line on ST2030 and ST2022. You can download here :
It working with Asterisk 1.2.24. It was based on different patch find on net (firmware 1.56).
5. Knowledge as to how to get dialing plan strings to work so that I do not have to press the OK button at the end of
entering the number so as to dial out...
Thomson: I don't know right now, but will be back to you ASAP.
AppTech: How soon is ASAP? I view this nowadays as a bug as the only part that works properely is the timer or someones
finger pressing the OK button. May as well not have a dial plan feature untill it works properely or is well documented so as
people know how to make it work.
Thomson: Dial plan synthax:
_To specify a_ _Enter the following_ _Result_
Digit 0 1 2 3 4 5 6 7 8 9 * Identifies a specific digit (do not use #)
Range [digit-digit]] Identifies any digit dialed that is included in the range
Range [digit,digit, digit]] Specifies a range as a comma separated list
Wildcard x x matches any single digit that is dialed
Wildcard . . matches an arbitrary number of digits
Timer T Indicates that an additional time out period of 4 seconds should take place before automatic dialing starts
AppTech: Ummm This info is sort of less usefull than the help link on the dial plan config page...
Just give us a few examples so that we know how to use it. The problem is that you think because
you gives us a programmers specification everyone will understand, not so. For example in the online help
it states "xxxx": If dial number exactly match 4 digits, then we send it to Call Agent.
and "xxxxT": If dial number exactly match 4 digits then we send it to Call Agent till timeout.
BUT I think the language is wrong. Maybe just the French to English is a problem. What actually happens
is that by adding a T into the dial plan segment, you force a delay of 2 seconds, not the value of dial-out timer
6. Call Parking - Ability to reposition/reorder soft keys? The issue for me is that my customers use the CALL PARK facility
all the time. So where do you hide it when in talk mode? ON PAGE 3! Given that when in conversation mode you do not have
a softkey assigned to the left button, can we have a way of shifting the CALL PARK to that position. The other odd thing is that
given you believe Call Parking is quite (correctly) important enough to have a web and APS field for it, then you hide the PARK
softkey when the phone is being used!!! Dumbfounded I am.
Thomson: I know that our softkey positions are not the best one. We can find altogether what is the best setup.
ST2030_Softkeys AppTech: They need to keep a function in the same spot i.e. "T r a n s f" moves around...
Ronald: Apptech: do you get the park function working with Asterisk? Mine is trying to do a park, but the call stays connected.
in addition to that: does anyone gets the pickup function work with Asterisk?
AppTech: Nope, they know it's broken in a few different ways but it is not a priority I guess
Call parking on this phone is brain dead. They enforce a "extn@hostIP" format which will never work
with Asterisk as it wants a DTMF of "#70". Seems the French don't use call parking much.
Overall the phone has some nice features but the competition is also strong so if Thomson want to have a solid Enterprise phone
they would be well advised to release good documentation and listen to their customers - People like us! -AppTech
Thomson: I try to do it.
Thomson: You are welcome. We have difficulties to gather some valuable feedback from our Asterisk clients are most of
them are not direct customers of us. Maybe this place is the best one to define the future of our products.
- Make headset feature working (now when pressing headset in 1.42 the tone is played on the horn, not on the headset)
- Ability to let the phone do a request to a URL (on incoming / connected / hangup calls). Also the DID and CID must be sent through
Thomson: I might not get it, but for me this kind of things are better handeled in the PBX
packetnet: I agree with Arian, the phone should be able to perform a lookup in the remote phone
book based on the incoming callerid and provide the name as a result.
- Like the remote phone book we will also have the feature to get the dialer name from a company web site. Maybe it can
Thomson: I also think the PBX is a better place for this one. Then it inserts it in the display param of the INVITE.
- Convertor from rj-9 headset plug to standard dual 3.5inch plug for computer headset
2. Making or answering calls with BLF keys instead of picking up the handset
When an incoming call rings, picking up the handset bridges your own extension to the incoming call.
For small call centers, it would be nice if agents had the opportunity to ignore incoming calls (as someone else could
deal with them) and still use their own phone for outgoing calls. Separating picking up the handset from answering a call, using dedicated BLF keys would then be the mandatory way to answer incoming calls or start new outgoing ones : to answer a call, you must press the blinking BLF key. To make a new call, you press another BLF key and get dialtone.
3. Directed pickup for monitored on non-monitored extensions
To be able to specify which incoming call to pick up when 2 extensions are ringing at the same time.
Pickup softkey seems ineffective today so as BLF keys : Pickup softkey today sends SUBSCRIBE messages which are not treated as Pickup demands by Asterisk. Maybe adding a configurable prefix (*8 or something like that) would allow us to tweak Asterisk to implement directed pickups.
6. RCTP XR
Add RCTP XR (RTCP extended reporting for call quality monitoring) as this feature is more and more often embedded in softphones.
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