| # |
Name |
Wish |
|
Rating |
| 1 |
Pieter |
Streamline the interface - get rid of separators & empty space, use buttons instead! (Call Dialpad Contacts) |
|
6990 |
 |
| 2 |
Vince |
Option to load LDAP contacts |
|
5209 |
|
| 3 |
Torsten |
Command switch to explicitly start video call to a given number. (e.g. "microsip.exe number /video") |
|
2320 |
|
| 4 |
tester |
Hotkeys, for example: "M" to mute microphone, "S" to mute speaker, "Up Arrow" to increase the volume... etc. |
|
2317 |
|
| 5 |
Max |
Video resolution setting. |
|
1925 |
|
| 6 |
Anon |
Add IPv6 support |
|
1860 |
|
| 7 |
tester |
a few functions like: redial, conference, do not disturb, call waiting, forwarding... etc. |
|
1797 |
|
| 8 |
Sega |
MS Outlook Address Book integration to the Contacts. |
|
1619 |
|
| 9 |
Nikola |
Possibility to switch among NICs if several physical ethernet interfaces are presented in the system |
|
1549 |
|
| 10 |
Max |
Fullscreen mode for video window (with upscalling). |
|
1532 |
|
| 11 |
tester |
buttons to activate/deactivate Echo Cancellation, Noise Reduction and A.G.C. |
|
1506 |
|
| 12 |
Netouch |
Simple API, or remote control API to integrate in existing infrastructure. Maybe listening TCP port, and receive commands, like call xxxxxxxxxxxxx, endcall, e.t.c. |
|
1506 |
|
| 13 |
Chris F |
Date format in 'Calls' tab is back-to-front. Should be picked up from Windows regional settings, or if that not possible should be configurable. |
|
1444 |
|
| 14 |
Jörg |
Add webRTC echo canceller like csipsimple did: http://code.google.com/p/csipsimple/source/detail?r=995 |
|
1321 |
|
| 15 |
a t |
http://www.freephoneline.ca VoiceMail Recording!
Seems like i can dial *98 but not hear my voice?
Anyways probably a codec issue with the message.
I like the small size, please make a usb distro,
Then i can just boot a key & just make my calls. |
|
1318 |
|
| 16 |
retset |
option to pause sending video stream and even completely stops it without call ends (start/stop/pause video during a call) |
|
1282 |
|
| 17 |
tester |
support for traffic prioritization, allow QoS signaling, by TOS value, DSCP, service type... |
|
1276 |
|
| 18 |
tester |
Desktop Sharing |
|
1244 |
|
| 19 |
retset |
option to hot-swap video source without cut the call |
|
1225 |
|
| 20 |
Anon |
Add NAT-PMP or UPnP support as STUN sometimes doesn't work well. |
|
1221 |
|
| 21 |
retset |
allow asimetrical bandwidth & video communication e.g. sending h264 and receiving h263, add option to completely disable sending a specific video compression (only at sending, not on receiving) |
|
1188 |
|
| 22 |
User |
Dial and caller ID quick log viewer |
|
1152 |
|
| 23 |
Pieter |
'Blacklisting' numbers for inbound calls - straight to voicemail |
|
1143 |
|
| 24 |
Knut |
Multiple sip accounts (at least 2) and the possibility to specify for each contact which sip account should be used |
|
1105 |
|
| 25 |
Beat We. |
Contacts: Option to define Groups |
|
1100 |
|
| 26 |
Supporter |
Merging Dialpad/ Contacts buttons into one. The label shown would be for the inactive selection. |
|
1089 |
|
| 27 |
Earl |
Would like to see a sister program: MicroXMPP for Jabber protocol. Should include ZRTP with SASL confirmation similar to JITSI. |
|
1088 |
|
| 28 |
Sam |
If you have 2 or more calls, switch to conference with all callers. |
|
1083 |
|
| 29 |
Machi |
Play sound when instant message receive and if message window not on top. Show message window on task bar. I never know if anybody send me a message if i have open another program. tnx |
|
1075 |
|
| 30 |
Mark |
area to change status message |
|
1073 |
|
| 31 |
Keyvan Tardast |
great little devil ^_^ please keep GUI simple. and few steps away from perfect ***
1- auto hangup (and perhaps auto close the call window)(and you can move this window in 3rd tab of main window) ***
2- PC speakers ring (good for ppl away from PC)***
3- |
|
1072 |
|
| 32 |
Beat We. |
Option to define a default outside line prefix (e.g. 0) to get an outside line, which is added automatically to all entered numbers > len(x) where x in our case would be 5 digits |
|
1054 |
|
| 33 |
tester |
sending and receiving files of unlimited size |
|
1052 |
|
| 34 |
cAroed |
group calls |
|
1037 |
|
| 35 |
carlos |
"Push to Talk" aka "Press To Transmit", very useful on multi-party conferences (microphone permanently muted is only enabled while pressing a key: usually "Ctrl") the only method 100% effective to avoid echoes and other unwanted noises |
|
1028 |
|
| 36 |
Stu E |
Add an automatic reconnect option for calls that drop for some reason. |
|
1015 |
|
| 37 |
whisher |
Please add hotkey from all MicroSIP windows (messages, local video, remote video....etc. ) by pressing "ESC" at keyboard: calls must end |
|
1011 |
|
| 38 |
Roman |
Add option for developers: print pure sip incoming/outgoing sip messages to message window during call. |
|
1008 |
|
| 39 |
Sushil |
Support of Require:100rel in 180 message (fully provisional acknowledgment support) |
|
1003 |
|
| 40 |
lol |
please make it possible to passthrough call data to a commodore c64 so finally i can log my calls. :) |
|
1001 |
|
| 41 |
Paul O |
Default audio hardware selection mode to choose Windows Default Communication Devices over Default playback/recording devices. http://tinyurl.com/7lcucuq |
|
996 |
|
| 42 |
asioboy |
asio4microsip
please read !
http://pastebin.com/mSi5R5wA |
|
993 |
|
| 43 |
retset |
option to view local or remote video during a call (or picture in picture with local video of small size on a corner) |
|
986 |
|
| 44 |
Brett |
Allow null audio device to be set for pjsip (-99). Useful if PC does not have a microphone plugged in and you just need audio output for PA applications. |
|
983 |
|
| 45 |
StanleyV |
Adding command line for the name and location of the configuration file "microsip.ini". Example 1:
MicroSIP.exe /i:microsip402.ini. Example 2:
MicroSIP.exe /i:folder402\microsip.ini. MicroSIP I use for testing multiple SIP accounts and is fantastic! |
|
982 |
|
| 46 |
Alain |
Support of picking up another ringing extension.
e.g.: by sending *8# |
|
976 |
|
| 47 |
Axel |
Send DTMF with command line parameter. We need this to control Sipgate Team a web-based voip solution. |
|
975 |
|
| 48 |
RE: tester |
RE: sending and receiving files of unlimited size (generate a .torrent and use the generated file with a regular bit-torrent client, using a tracker or the DHT method) |
|
974 |
|
| 49 |
zsll2 |
Option to disable focus stealing of action (IM) window when auto answer is on. |
|
973 |
|
| 50 |
choices |
please add option to set framerate for sending video, 10fps,15fps or 18fps is enough, and it saves bandwidth, very useful for slow internet lines like mine, setting video frame size is also desirable |
|
970 |
|
| 51 |
Julian |
Unregister from SIP server/Away/Redirect to voicemail when Windows screen saver is on or lock screen has been activated. |
|
968 |
|
| 52 |
RE: grommir |
record calls as SPX which can be played with VLC which is used by everyone. We use SPEEX to optimize VoIP calls so why not for recording as well?! see the difference in size with same speech quality hackerpublicradio.org/syndication.php |
|
955 |
|
| 53 |
Dominic |
Great project, thanks so much! But would love distinctive ringing please! So you can have different ringtones for different contacts. |
|
955 |
|
| 54 |
Vince |
Any kind of message when you've missed a call (i.e. a flashing icon or something). |
|
949 |
|
| 55 |
Giuseppe |
Flag to remove ringtone |
|
928 |
|
| 56 |
Khoa Pham |
Conference call |
|
926 |
|
| 57 |
Vasily |
A way to transfer calls. Blind transfer is very old fashioned and not very useful these days.
More friendly interface. |
|
909 |
|
| 58 |
whisher |
please, add minimize button on messages window, please remove close button from it, when I want to get away from the midst that annoying window for me, communication is cut off |
|
907 |
|
| 59 |
Jared Savage |
Alternative ringer if already in a call (silent?) |
|
884 |
|
| 60 |
Ryan |
Be able to auto answer only specific extensions, so that intercom functionality is available without auto-answering outside calls |
|
882 |
|
| 61 |
amix |
Use Windows contacts. Should solve importing from Google, etc., too, since it can act as a bridge. |
|
871 |
|
| 62 |
el_es |
2. Linux version for GTK or Qt desktop ... |
|
866 |
|
| 63 |
Khoa Pham |
Support ICE and TURN |
|
856 |
|
| 64 |
MIT |
Hide Phone Number When Making a Call |
|
856 |
|
| 65 |
Druify |
Improved screenreader accessibility: Edit boxes assigned to correct labels in Account dialog; Apart from suggested hotkeys, possibility to reach any object with tab/shift tab for dialog elements and ctrl+tab/ctrl+shift+tab for switching tab controls. |
|
855 |
|
| 66 |
Mochammad Effendi |
I need group contact, like Family, Fren, Work etc.
Thanks
ME |
|
854 |
|
| 67 |
Bars |
Automatic detection and switching audio when connecting a Bluetooth headset, the other headset, and the same - to the first in the list of audio devices when you disconnect. List of priority of audio devices to use depending on their avaliability. |
|
851 |
|
| 68 |
Jose |
Ability to select contacts from a dropdown menu or similar from call transfer dialog. |
|
848 |
|
| 69 |
vxed |
Option for disable history of calls |
|
841 |
|
| 70 |
tim |
Add G723.1 |
|
840 |
|
| 71 |
rus7 |
Please, make option "auto answer" as follows: auto answer after N calls. |
|
839 |
|
| 72 |
Lucasinc |
Command switches to pick-up incoming call |
|
825 |
|
| 73 |
Michal P |
Make Windows 8 Modern UI version (and make simple like current version), dont make it for free, be the first in the store and be rich my dear developer! (your first customer) |
|
824 |
|
| 74 |
anon |
Close to tray. I have reverted back to 3.3.14 because it's too easy to exit the phone now. Minimize button is nice but why change the close button behavior? |
|
824 |
|
| 75 |
Pero |
under account settings, additional parameter for setting re-register time in seconds (default=300s). |
|
824 |
|
| 76 |
Mit |
I Would like to have Media Encryption with RC4 or other Types. |
|
822 |
|
| 77 |
Roman |
Multiple SIP accounts active in parallel |
|
813 |
|
| 78 |
booBot |
ZRTP (RFC 6189) support for call encryption. |
|
810 |
|
| 79 |
tim |
Add audio codec G723.1 |
|
809 |
|
| 80 |
Shaun |
Snap windows for resize and positioning (to themselves and edge of screen). |
|
805 |
|
| 81 |
frank |
do not disturb DND feature support |
|
805 |
|
| 82 |
Xsmael |
Run an executable on incomming call, and pass arguments to it(id,name,phone number...). You can also improve it by adding "events" managment, i mean, describe some events and let the user chose a program to launch whenever this event occurs, again some pa |
|
802 |
|
| 83 |
darkgerion |
Archiving messages (file) |
|
801 |
|
| 84 |
Marce Cz |
Enable "Skins" option for MicroSIP to change the layout. |
|
795 |
|
| 85 |
Bobby |
Highwe bitrate codecs like FLAC or OGG for high quality audio between two softphones. Reason: Radio station remotes. |
|
794 |
|
| 86 |
Christian |
Add some version information (possible locations: bottom of the settings page or maybe add a separate about dialog) |
|
794 |
|
| 87 |
Youri |
Please, add G726 codec. It is in PJSIP lib/plugins section. If you need in reference implementation here is the source codes:
http://www.codeforge.com/dlpre/8930/45d6badd00df.rar__.html |
|
777 |
|
| 88 |
Robo |
Notify missed calls, change tray icon when have missed call, flash icon or display number of missed calls (like PopTray). |
|
777 |
|
| 89 |
Toha |
Save contact from the number in call list |
|
775 |
|
| 90 |
Joop |
Add a column "Contact" to the calls list. This shows the name from the contacts list if it is found. |
|
763 |
|
| 91 |
Lelio |
When in single call mode, allow to hangup via command line with "Microsip.exe /h"; also allow to place an "hangup and then call" command like "Microsip.exe /h 12345".
A possible fix at http://www.leliodellapietra.it/microsip.txt |
|
754 |
|
| 92 |
Steven |
Option to allow DTMF tones to last for a certain length of time, useful for navigating options/menus when call quality is low. Optionally tie tone length to press and release of keys/mouse. |
|
748 |
|
| 93 |
Stefano |
Bluetooth management.
Actually if connected a BT headset, the audio channel rimains active or you have to manage it manually.
It could be useful to activate it only when needed (as Bluesoleil BT stack does). |
|
748 |
|
| 94 |
salaros |
Add the possibility to prepend prefix to ANY number before actually calling it. In my case it's '0'. i.e. 800900800 => 0800900800
P.S. with the exception of "internal" numbers (other PBX users on the network), maybe via an exception list. |
|
747 |
|
| 95 |
frank |
support BLF (MIME type dialog-info accordingly to RFC 4235),A patch had made by Максим Кондратенко to pjsip. |
|
735 |
|
| 96 |
Mario |
Can be useful to have an AnswerMachine that auto answer and play a wave file at the phone caller, for example a welcome message with informations and after the message caller can record a message.
Thank you for this stable and well done software! |
|
733 |
|
| 97 |
Riaan Olivier |
Possibility to register against a specific VLAN ID and QoS number. require softphone to be given its own IP address |
|
733 |
|
| 98 |
Pierre |
Put Option to show or not Incomming call Window. With cmdIncommingCall give caller Name plz |
|
727 |
|
| 99 |
Bob |
Can you include Opus codec for high quality stereo transmission? |
|
724 |
|
| 100 |
zth |
support for snapshot of remote in a call,and save it as .jpg |
|
724 |
|
| 101 |
DSJ |
Please create Metro app in Windows 8 (arm surface).
See the design of csipsimple tablet version. |
|
723 |
|
| 102 |
Yuz |
Miss Call notification in system tray. |
|
718 |
|
| 103 |
Andrei |
record calls NOT as mp3, but AS IS, encoded with used codec (gsm, Speex etc.) |
|
711 |
|
| 104 |
Pedja |
Is it possible to have some kind of password when MicroSIP is set to autoanswer mode? I would like to be able to limit calls to autoanswer MicroSIP ony to few known people. |
|
704 |
|
| 105 |
frank |
dynamic switch from different input devices when in video call session. e.g from color bar to usb camera or vice versa . |
|
693 |
|
| 106 |
Guy |
implement RCC (Remote Call control) features for use with eg bluetooth connected Jabra headsets |
|
693 |
|
| 107 |
tommy |
support for snapshot to jpg |
|
686 |
|
| 108 |
Werner |
Ability to past numbers with dashes and parentheses. I tried to paste a number like 06-12345678, but nothing happened. Until I removed the dash and tried to copy-paste again. Please add a filter to remove all non numeric characters (except +) when pasting |
|
684 |
|
| 109 |
Cat |
Recognize the header 'Reason: SIP ;cause=200 ;text="Call completed elsewhere"'.
When a call to MicroSip is canceled because the call is answered by an other client we still get a "call aborted" status in MicroSips call list. Asterisk and others send it. |
|
678 |
|
| 110 |
Christian |
Update call log from address book: If you add a number from the call log to the address book, replace all entries of that number by the address book entry (number -> name) |
|
673 |
|
| 111 |
frank |
MWI voice mail feature support.and maybe video mail in the feature. |
|
663 |
|
| 112 |
Ray |
Set RTP port please. |
|
658 |
|
| 113 |
Guillaume |
There should be a very visible "you have voicemail" indicator. |
|
651 |
|
| 114 |
Evgeny |
Please add RTCP-XR support |
|
644 |
|
| 115 |
mv |
Playback of remote DTMF tones during call.
(These are often used as confirmation to IVR actions and so on) |
|
633 |
|
| 116 |
jason |
complete contact URI in 200ok,It is now sip:ip:port,
should be sip:user:ip:port. |
|
628 |
|
| 117 |
Ken |
Better handling of “speaker” and “microphone” devices to automatically use Bluetooth headset when connected and speakers when not connected. |
|
628 |
|
| 118 |
Refrain |
VP8 or VP9 video codec.
https://en.wikipedia.org/wiki/VP9
https://en.wikipedia.org/wiki/VP8 |
|
620 |
|
| 119 |
Gloria |
At least for me is almost perfect. You could add a call recorder and maybe an option for put a custom on-hold music xd. tx |
|
620 |
|
| 120 |
R |
Is it possible to specify the directory XML MicroSIP disable checkbox "Presence subscription" for each contact? |
|
618 |
|
| 121 |
Carlo |
optional "incoming call window in center of screen" |
|
616 |
|
| 122 |
cbm |
ZRTP please |
|
613 |
|
| 123 |
Adrian |
Can't see any option after installation to make microsip start automatically when windows starts. |
|
613 |
|
| 124 |
Bill |
How to Disable Contacts tab or the messaging functionality. |
|
610 |
|
| 125 |
Marcio |
Incoming text message sound is the feature I mostly miss on MicroSIP. Intranet text messages are one of the main uses of MicroSIP here, and we miss very much a sound that signals when a new text message has come. |
|
607 |
|
| 126 |
Vlad |
CallerID copy in clipboard. |
|
604 |
|
| 127 |
Alex |
Allow option to prevent the main window from popping up when calls are received. The window should remain unseen the entire session. |
|
604 |
|
| 128 |
Andreas M. |
Use the Windows Contacts (or equivalent) API for addressbook. Autocomplete names or numbers from there. |
|
599 |
|
| 129 |
tomash |
auto provisioning via http |
|
598 |
|
| 130 |
Johen |
Option to start multiple MicroSIP instances for testing some features across SIP-server. |
|
595 |
|
| 131 |
Travis |
Already been said, but recording calls into audio file. Any audio file that is not too large file size. Contact me on my email, I will be willing to offer you money for your time to implement this. |
|
589 |
|
| 132 |
Bipin |
level indicators for mic and speaker |
|
587 |
|
| 133 |
tester |
2 essentials missing features to have is an automatic audio recording or a toggle button to start record in a call. Also I would like to have a little sound wave bar display on the Dialpad for the Microphone and Sound so I can see and know the audio is ac |
|
585 |
|
| 134 |
Alex |
Please make it possible to visualize missed calls. (Windows 7) |
|
582 |
|
| 135 |
Francisco |
Option to import and export contacts. |
|
581 |
|
| 136 |
Simon |
The incoming call pop-up window to be 'always on top' like the main application window can be. At current it gets lost behind my active window. :( |
|
581 |
|
| 137 |
telephoner |
Place a note on the webpage that "With WineBottler 1.7.16 from http://winebottler.kronenberg.org/ MicroSIP does actually also work on Intel based Macs." – Even if you don't grant any support and don't guarantee for anything. |
|
580 |
|
| 138 |
telephoner |
Full DTMF keypad including DTMF-tones A, B, C and D http://en.wikipedia.org/wiki/Dual-tone_multi-frequency_signaling |
|
578 |
|
| 139 |
Martin C |
Support for the Call-Info Header as per RFC3261 section 20.9
I would be nice so we can send a JPG url to the phone to display it before answering the call. |
|
570 |
|
| 140 |
Yann |
Colours :)
- GREEN = Answer & Call
- RED = Decline
(Eventually icons for the same buttons) |
|
562 |
|
| 141 |
Francisco |
Option to make conferences like eyebean with or without video. |
|
558 |
|
| 142 |
Knut |
Configure which RDP ports to use |
|
557 |
|
| 143 |
Hassan |
The ability to customize the packetization time (ptime), e.g. 10ms, 20ms, 30ms. |
|
557 |
|
| 144 |
Refrain |
Settings:start the program minimized to tray. |
|
555 |
|
| 145 |
Francisco |
Button to access camera settings. |
|
553 |
|
| 146 |
Bago |
Lookup incoming phone number on tellows and/or phone number to name lookup services (or maybe the excerpt of the first result of a google search for that number). |
|
552 |
|
| 147 |
marcin |
I wish that the program could be set the maximum number of incoming calls, eg 3,4,5 maximum at one time. The radio stations when broadcast live at one point calling 50 people, the program crashes. It's nice to be able to when this limit. Thank you very mu |
|
545 |
|
| 148 |
el_es |
3. Have the wish entry window count characters remaining ;) |
|
545 |
|
| 149 |
roman from yandex.ru |
We use Microsip as contanct center agent softphone. ~ 3000 agent. 3rd party call coltrol has been done in microsip, but it is not standar metod. Need bria metod to automatic answer call from SIP server, it is standart de facto. |
|
542 |
|
| 150 |
chrml |
import/export vCard format (vcf)
https://tools.ietf.org/html/rfc6350 |
|
537 |
|
| 151 |
kk |
Add option to change key translated to #, and work it in input field |
|
536 |
|
| 152 |
zane |
call with a predial injection like S0 |
|
533 |
|
| 153 |
el_es |
4. Option for the CallerID display to remain visible (with timeout or not) if other device of the same extension picked up the call; Button to copy the incoming CallerID to clipboard. |
|
531 |
|
| 154 |
Sascha |
1s pause DTMF tone. By ',' or 'p' delimiter for example. Would be awesome! |
|
530 |
|
| 155 |
Wesley Chapman |
Allow user to configure different code than ** for call pickup code and group pickup code. In our implementation the code is *8 + ext. |
|
528 |
|
| 156 |
Carlos |
Add option to "Register every X segonds" in account properties |
|
527 |
|
| 157 |
Alex |
Changing Soundcards without disconnecting. |
|
526 |
|
| 158 |
Tommy M. |
10 digit dialing. configurable default country code, etc. |
|
526 |
|
| 159 |
loukas |
DND option |
|
525 |
|
| 160 |
Matt Welborn |
Allow clickable URLs in SIP Messaging. I'd like to be able to send links via SIP Messaging to the call window and have them clickable and open in the default browser. |
|
523 |
|
| 161 |
Suraj |
keyboard shortcut for microphone mute. mute button to go red when enabled. mini interface (always on top) with mute, speaker and hangup buttons. |
|
522 |
|
| 162 |
FL |
More security: 1. Optionally, the account password could be entered at application startup time. 2. A password to lock the phone. |
|
515 |
|
| 163 |
Artem |
COM object to use Microsip in other applications |
|
512 |
|
| 164 |
Mike |
Please make "close to tray" configurable. I hate the systray full of icons of running applications but I get that there are people who prefer it that way. Simple option in ini would be sufficient, no UI needed. ;) |
|
512 |
|
| 165 |
el_es |
1. SIP Originate : tell your Asterisk PBX to ring your desk phone, you pick up the handset and then the PBX dials out to the number, on behalf of your extension; Like SIPTAPI but without dialer.exe |
|
511 |
|
| 166 |
Alex |
Ability to register multiple accounts concurrently for use with platforms such as Vicidial which may be composed of multiple asterisk servers. |
|
509 |
|
| 167 |
Peter |
TLS keypinning |
|
505 |
|
| 168 |
Norbert |
Import contacts from vCard and csv, maybe directly from google. The only thing that prevents me from using microsip for now... |
|
504 |
|
| 169 |
Michael Schelin |
Text Messaging - Vertical list for message phone numbers (instead of horizontal tabs) |
|
501 |
|
| 170 |
adrian |
Option to hide offline contacts |
|
500 |
|
| 171 |
Tim |
open a URL when there is an incoming call.
Example www.example.com/contact.ph?number=31523234 |
|
496 |
|
| 172 |
Hassan |
Add a button to switch from one codec to another during the call e.g. from G711 to G729 if connection starts going down, then MicroSIP should send a re-INVITE to the remote party with the new codec. |
|
486 |
|
| 173 |
alvin martin |
notice in icon tray when a chat came in or a call you missed. Like maybe a color change from blue to green or red. |
|
486 |
|
| 174 |
alvin |
have Ringing Sound setting like this ..\MicroSIP\ringin.wav so you do not have to keep making this setting when you use this phone portable. |
|
483 |
|
| 175 |
Marius |
Group contacts by groups.
PS. cantact email has to few characters. |
|
476 |
|
| 176 |
Jon |
When Make Active is checked, would be nice if the client automatically reactivated/reregisters if the network connection is lost and resumed again (e.g. resuming from sleep, etc.). Currently, I have to toggle this manually. |
|
469 |
|
| 177 |
Alex |
Add Description/Tag field to accounts just for identification purposes. Very use full if keep switching accounts all the time.
Thank you. Your effords are greatly appreciated. |
|
465 |
|
| 178 |
Jan Sebosik |
Option to specify portrange for rtp/UDP (and maybe also for SIP) |
|
449 |
|
| 179 |
Jason |
Use account name instead of sip url in the menu so that it's straight-forward which account to switch to when there are multiple accounts |
|
443 |
|
| 180 |
Andy |
"hangup" Parameter over command line (CMD) |
|
437 |
|
| 181 |
Pavel |
Option to disable only local DTMF sounds when dialing. |
|
437 |
|
| 182 |
Johan |
An option to set automatic updates for new stable versions. |
|
437 |
|
| 183 |
Goran |
Configurable User Agent string (just like CSipSimple for android allows).
Lots of VoIP Service Providers only allow certain User Agents using a whitelist. |
|
436 |
|
| 184 |
Yann |
Loud speaker button : a button that will output the sound to the "ringing device". This way you can put your converstaion on loud speaker (when waiting for a person to asnwer or doing a conf call with others in the room). Please ad it in the "Tab view of |
|
429 |
|
| 185 |
Yuri |
Option for window "Incoming Call" behavior. Since 3.10.2 it always popup. Before was more predictable: Popup if minimized, flash bar if has no focus |
|
418 |
|
| 186 |
Vadim |
XCAP for Contact list Load/Save |
|
414 |
|
| 187 |
Davie J |
Auto provisioning server, have a central server and use codes or keys to provision clients |
|
414 |
|
| 188 |
Sergei |
Support USB SkypePhone (Buttons, Ringer) |
|
399 |
|
| 189 |
MZYChina |
Define DTMF digits sends as in-band DTMF,or can be selected. |
|
394 |
|
| 190 |
myself |
This MicroSIP phone has no instructions for setting it up, nothing, just technical garble on the help page. The first thing it says on the help page is this: "Your account SIP server." WHAT about an account SIP server? Why are there no instruction anywher |
|
382 |
|
| 191 |
Darko |
VLAN tagging. It should have virtual NIC built in that you can put on different VLAN. This way it segregate the traffic between data and voice. This is possible with physical voip phones. It should be possible with software phones (SIP) clients as well. I |
|
380 |
|
| 192 |
Xavier |
Assignable shortcut to answer/hangup a call, e.g. defined to answer call with headset button play/pause. |
|
373 |
|
| 193 |
P M |
Regarding User State. I would like to see a drop-down box that highlights our user state, so that, we would be able to see if we are Available, in Training, on Break, etc. Currently, when the status is changed, it only shows at the bottom of the page and |
|
368 |
|
| 194 |
walter |
add an option that allows you to block access to the configuration parameters and account configuration to limit the configuration of these things only to administrators who have the unlock password |
|
364 |
|
| 195 |
Lospejos |
Plugins API support to be able to make plugins - f.e. events handler (inbound call, outbound call, status changed) etc. |
|
351 |
|
| 196 |
Jim Hunt |
Ability to play an audio file such as music or a spoken message over the line to a caller when a call is placed on hold. Example of file could be hold.wav. |
|
343 |
|
| 197 |
Hoang Nguyen |
Group chat |
|
333 |
|
| 198 |
Однократный зумер |
MicroSIP-3.11.0 при получении вызова происходит однократный звонок и вызов остается а звука нет. Откатил на предыдущую версию - там все нормально. |
|
331 |
|
| 199 |
MAX |
ability to execute the cmdIncomingCall also on rejected call |
|
327 |
|
| 200 |
florian |
Settings-Option to deactivate the auto-away feature. |
|
324 |
|
| 201 |
Alexis G. |
Possibility to choose Default Communication Device or Default Device (ie Default for Ringing and Default Com for Call Sound And Mic) |
|
315 |
|
| 202 |
GF |
Replace strings in the numbers before actually calling to fix special characters that cause problems:
"+49(0)" -> "0"; or
"/" -> "" |
|
307 |
|
| 203 |
atz |
Auto-Answer (All calls) button switch on dialpad panel |
|
305 |
|
| 204 |
PRIHLOP |
record calls as amr |
|
302 |
|
| 205 |
Benjamin |
adding silent installation parameters, multi windows account configuration etc... thanks |
|
302 |
|
| 206 |
michael |
Bug with SIP |
INVITE SDP:
m=audio 4004 RTP/AVP 8 0 18 101
m=video 4008 RTP/AVP 99 98
|
200 OK SDP:
m=audio 4004 RTP/AVP 8 0 18 101
|
But SIP RFC must:
m=audio 4004 RTP/AVP 8 0 18 101
m=video 0 RTP/AVP 99 98 |
|
297 |
|
| 207 |
Supafly |
Hovering display window in the corner of the screen with caller ID, call duration and hangup button. |
|
293 |
|
| 208 |
Elis |
Disable ring-window on top |
|
281 |
|
| 209 |
Isaac |
I hope this software would be good |
|
277 |
|
| 210 |
ar |
Please add MPEG Layer 2 codec for high quality audio. With MPEG Layer 2, you can connect to a IP Codec device for broadcasting and transfer high quality audio between remote locations or studios. There is no freeware softphone that supports MPEG layer 2. |
|
272 |
|
| 211 |
Boris |
Ability to change font size for call logs and contacts (current font is too small). |
|
265 |
|
| 212 |
Badiuz |
Hotkeys to answer a call like press enter without move a cursor and click button anwser |
|
263 |
|
| 213 |
walter |
See the version number of microsip : Ex "About menù" with all the information of microsip. |
|
259 |
|
| 214 |
Raul |
Option to disable (or to reduce the volume) of the beep sound in the ending of a call. |
|
259 |
|
| 215 |
AndreyParkhomets |
Fix IP selection when VPN used, otherwise no incoming voice |
|
258 |
|
| 216 |
walter |
A configuration flag that eliminates any sound signals in headphones for events (ES: autoanswer beep) |
|
254 |
|
| 217 |
walter |
An option to hide numeric dialpad |
|
251 |
|
| 218 |
thefish |
Built-in OpenVPN client like Yealink phones |
|
251 |
|
| 219 |
Adriano |
Option to select/remove columns in the contacts/history tabs. |
|
231 |
|
| 220 |
Andre |
Display won't sleep when microsip is running. "powercfg -requests" lists the "microsip.exe" process as the culprit (under "DISPLAY").
Rebooting or restarting microsip does not correct correct the problem. |
|
224 |
|
| 221 |
highbaser |
Don't streamline the interface - Please don't ruin MicroSIP with today's crappy UI design standards (Material, Metro, etc.) |
|
219 |
|
| 222 |
Yu |
Hi, could codecs [email protected] or G.729a be added? |
|
216 |
|
| 223 |
Sandor |
Option to load CARDDAV contacts |
|
216 |
|
| 224 |
Yvy |
Could you please add codec G.726-40 and G.726-32? (G.726 is used to convert G.711A/μ to and from 40 or 32 kbps channels in DCME.) |
|
210 |
|
| 225 |
Badiuz |
settings to add dial prefix automatically when calling contact |
|
207 |
|
| 226 |
Matthew |
Automatically unmute mic once call is complete. |
|
207 |
|
| 227 |
memo |
add a speaker phone facility (second sound card) |
|
205 |
|
| 228 |
zwostaa |
Set focus on incoming call window / bring it to foreground when ringing (OS: windows 7) |
|
203 |
|
| 229 |
Kaile |
it doesnt work correctly if connection built with vpn, the pbx is located in different location, and i want to use MicroSIP in different location and there is site-to-site vpn, but it doesnt work correctly. |
|
203 |
|
| 230 |
ACL |
Why don't you use paypal? |
|
200 |
|
| 231 |
JMM |
Avoir une version microsip sans possibilité d'enregistrer les appels (comme cela existait dans la version 3.19.8. |
|
199 |
|
| 232 |
Oleg |
Add an option like in X-Lite "Play DTMF tones back to me", or make DTMF quieter (10% - 20% - 30%......100%) |
|
199 |
|
| 233 |
Yu |
Could you add an indicator that shows which codec is actually being used during a call? |
|
198 |
|
| 234 |
Pedro |
Add TRANSFER and CONFERENCE buttons to main window and to Single Call Mode |
|
197 |
|
| 235 |
Sergejj |
Добавьте пожалуйста регулировку громкости "Звуковых событий" в меню "Настройки" . Во время набора номера звук в 10 раз громче чем ответ телефонной станции. Звонить не комфортно. приходится отключать полностью "Звуковые события". |
|
193 |
|
| 236 |
Pablo |
Dial *76 (or custom) when click DND on the phone to let asterisk know |
|
189 |
|
| 237 |
Joel |
When existing a contact and going back to the contact, previous messages do not appear |
|
186 |
|
| 238 |
admin |
receive http api get requests - e.g. to open the door (DTMF), switch video, play remote audio message ... |
|
182 |
|
| 239 |
Rudflodur |
Extract invalid characters from number (e.g. 0123/456), other characters like (0123.456) works.
Writing phonenumbers with slash is common in Germany |
|
175 |
|
| 240 |
Jameel |
Option to add prefix to every outgoing call. thank you |
|
174 |
|
| 241 |
mu5tfind |
Call on hold reminder, where after X seconds/minutes a sound or beep is heard through the "speaker" or "ring device" of MicroSip. |
|
167 |
|
| 242 |
David |
Possibility to download call history in csv or xls file. |
|
165 |
|
| 243 |
VTRAORE |
TRAORE VASSOULEYMANA |
|
165 |
|
| 244 |
Ndi |
Multi Line and conference on single window |
|
163 |
|
| 245 |
Riccardo |
Try to reconnect every 'xxx' seconds if some error shows. Appreciated when you use slow dhcp and microsip on startup, if windows doesn't get quickly an ip microsip stays offline and you have to reconnect it manually. |
|
162 |
|
| 246 |
Stefano |
lookup on Tellows, shouldianswer or other antispam services (after stripping anything after @ in the incoming caller identifier). |
|
158 |
|
| 247 |
David |
An OCX to integrate into other apps that return values such as: call time, call status, error codes and so on. |
|
158 |
|
| 248 |
Dimitri |
IAX protocol funcionality if possible. |
|
156 |
|
| 249 |
ali eminent |
Restriction on call recorder, so that other user can't disable call recordings or change the recording path. |
|
155 |
|
| 250 |
Marco |
Display "info" information in popup window when incoming call coming |
|
155 |
|
| 251 |
Julie |
Make dialpad buttons smaller, so overall width would become smaller too. And maybe make an option to hide them completely. |
|
154 |
|
| 252 |
Eric |
A notification for when someone leaves a voicemail |
|
153 |
|
| 253 |
tanguy |
Can use handset button to hang up / pick up ( Jabra handset ) |
|
152 |
|
| 254 |
Fabian |
Hi, it would be great if i could right click a call in "logs" to add it as "contact". |
|
151 |
|
| 255 |
Guillaume |
Option to upload contact from csv |
|
149 |
|
| 256 |
Blake Sinnett |
ZRTP support, and the ability to select the Opus frequency, channels and bitrate. |
|
147 |
|
| 257 |
list for auto answer |
Auto answer has many options, i don't know. I want a easy choice like "8499*@sip*.com" or blacklist whitelist. "Directory of users" is hard to set for new user. |
|
147 |
|
| 258 |
Mark |
I am looking for someone calls me, it is a telemarketer, and I dial a code, say *32 and it adds that number to a block list |
|
144 |
|
| 259 |
Mike |
Open url at incoming call |
|
143 |
|
| 260 |
Alex |
Setting in preferences for automatic conferrence mode Option, so that participans cann be directly part of a current call. |
|
141 |
|
| 261 |
calas |
Include non-ECC certificates or make website available via pure HTTP (not HTTPS). Inaccessible with older browsers on Windows XP. |
|
139 |
|
| 262 |
NGUYEN |
Search all accounts with same network. Use with RadminVPN, you can only call via IP, not via name |
|
139 |
|
| 263 |
Shadow373 |
Add Blacklist for microcip. |
|
138 |
|
| 264 |
Ron |
hAS IT EVERY OCCURRED TO ANYONE THAT THESE PEOPLE DO NOT CARE LOL, IT NEEDS A CRM/ DATABASE TO BE ATTRACTIVE TO MOST BUSINESSES |
|
137 |
|
| 265 |
Ramon |
PJSIP Support on port 5080 and SRV Record for registration |
|
135 |
|
| 266 |
Dmitry |
The password should not be displayed in GUI or the account settings may be able to be blocked for viewing by non-administrators |
|
134 |
|
| 267 |
Rickylee |
Can I modify manual and dynamic below rtpmap value ? like a example of AMR 12k SDP. I wanna change rtpmap from 120 -> 97 ,
m=audio 11706 RTP/AVP 121 120 101
c=IN IP4 192.168.2.147
b=TIAS:23850
a=rtcp:11707 IN IP4 192.168.2.147
a=sendrecv
a=rtpm |
|
133 |
|
| 268 |
Russ |
Add new account by importing from a config file, maybe achieved by associating a new file extension with the exe and a new command line switch |
|
133 |
|
| 269 |
Abe |
Ability to receive multicast rtp streams (for paging) the way desk phones do |
|
133 |
|
| 270 |
Stu E |
add higheer bit rates for OPUS codec. as high as 128kbs would be nice |
|
132 |
|
| 271 |
Paralea |
Don't use a separate window for the calls. They get buried onscreen, and it's hard to find them to hang up. Use the actual window where you are dialing from to end calls. |
|
131 |
|
| 272 |
Evan |
Set outbound caller ID in settings. |
|
130 |
|
| 273 |
mu5tfind |
Audio confirmation when putting call on hold and bringing it back from hold. |
|
129 |
|
| 274 |
Michael |
Make it possible to send a donation by wire transfer to Bank-account, without paypal, credit-card etc. |
|
128 |
|
| 275 |
Blake Sinnett |
Add hotkeys for shortcuts (control+1-8) as well as for call management (control_h to hold, control_e to end call, control+t to transfer, etc) |
|
127 |
|
| 276 |
Dmitri |
Build for Linux/FreeBSD |
|
124 |
|
| 277 |
BPhilippi |
Advanced SIP parameters -
Registration timeout, re-register interval, sip session timers etc. |
|
123 |
|
| 278 |
Petra K. |
After dialling a number from an external program, MicroSIP gets automatically maximized. Could its dimensions remain the same at all times? |
|
122 |
|
| 279 |
Fabian |
Maybe you could consider to access external Phonebooks, like Fritz.box devices provide. |
|
122 |
|
| 280 |
Glukinho |
Allow to make calls without speakers/microphone/audio device at all, especially in RDP session. Useful to troubleshoot SIP connectivity issues, when you need to know quickly if a call can be dialed at all. |
|
120 |
|
| 281 |
Moishe |
Choose client IP address to bind to eth0, eth1 (for sip account) |
|
117 |
|
| 282 |
mu5tfind |
More than 8 configurable shortcuts |
|
113 |
|
| 283 |
Kevin |
The ability to support MLPP User Agent features. Would be very useful for testing some SIP networks. |
|
112 |
|
| 284 |
Mary |
Google Contacts SYNC (at least one-way Google -> MicroSIP) |
|
108 |
|
| 285 |
Alvaro José Peña |
Themes or Skin for GUI like xlite |
|
102 |
|
| 286 |
Ober |
Headset integration - Sennheiser HeadSetup |
|
102 |
|
| 287 |
George |
Server Certificate Validation when using TLS |
|
101 |
|
| 288 |
Thomas |
Routers like AVM have a simplified setup for VOIP accounts with a number of providers across Europe. A similar functionality within MicroSIP would be great (e.g. Vodafone, Telecom, 1&1, etc.) |
|
100 |
|
| 289 |
Michele |
It would be very comfortable if the "Call" button would turn into "Hang up" if there is a call in progress. |
|
98 |
|
| 290 |
Abe |
Use default communications devices in Windows audio system, so it automatically reduces other audio when a call is active. |
|
97 |
|
| 291 |
lucas |
1.Add option to "Register every X segonds" in account properties.
2.MWI voice mail feature support.and maybe video mail in the feature. |
|
97 |
|
| 292 |
TEL-PRO |
Ciemny motyw lub kolor skórki. |
|
94 |
|
| 293 |
SaiKalyan |
Could you please add prerecorded greetings internal audio play. auto-answer & recording option is already present so if this greeting & attendant option will be added it will be a fully functional answering machine for me. if any one can do this please c |
|
91 |
|
| 294 |
Filip |
Allow connecting to SIP account by clicking on link (similar to call:) |
|
89 |
|
| 295 |
Frank |
Higher quality Opus codecs, perhaps 48KHz and 64KHz for radio remote connections. Local account is great for direct connection to broadcast codec HW like Comrex! |
|
87 |
|
| 296 |
dman |
Separately route Call Ring to Speakers and Voice Audio to Headphones option Windows 10. |
|
86 |
|
| 297 |
Dawid |
Multiple phone lines/accounts with separate dialpad/control on single screen with separate speaker and microphone selection. |
|
83 |
|
| 298 |
Juanfran |
Make incoming call window not to steal focus. It is annoying when you are typing in some other window and the phone steals focus. If you disable the bring-to-front setting, you get a non-blocking OS notification but it lacks of answer/reject buttons. |
|
83 |
|
| 299 |
Charlie |
German translation and userdefind coulors/styles and
size |
|
82 |
|
| 300 |
depape |
multiple sip accounts registrations on-line |
|
82 |
|
| 301 |
Federico |
Add extensive log of voip protocol in order to troubleshoot voip issues |
|
80 |
|
| 302 |
Corporate User |
Call forwarding to a mobile number for afk usage |
|
80 |
|
| 303 |
Fischer |
Make it Citrix ready. It works great in XenApp, but all users have the same settings, because microsip.ini file stored on same location then exe file. e.g. microsip.exe is stored in C:\programms and microsip.ini the same. |
|
79 |
|
| 304 |
Youri |
The setting for Opus, "maxaveragebitrate=20000", means 8kHz mono bandwidth only, and G.722 sounds better. 50000 would make MicroSIP sound as good as PhonerLite! ;) |
|
78 |
|
| 305 |
MatthiasSchröder |
Map special media key (i.e. windows audio mute key) to end call for multimedia remote controls whithout stop button. |
|
77 |
|
| 306 |
Kems |
Add a transparency slider in the options to be used in combination with "always on top" |
|
77 |
|
| 307 |
Moshe |
Should work with siptapi |
|
76 |
|
| 308 |
Maxy |
Video format 16:9, not 4:3 :) |
|
76 |
|
| 309 |
Spidy |
I was wishing for uPnP.
Improved Messaging.
Above all, we wish you well !
Great Product !!! |
|
70 |
|
| 310 |
Kevin K |
Please add option to modify Payload type on Video codecs. |
|
69 |
|
| 311 |
natSIP |
add set CLI "Set(CALLERID(all)=xxxxxxxx)" in setting. |
|
68 |
|
| 312 |
Mos |
Possibility to use multiple outbound providers |
|
67 |
|
| 313 |
JT |
Add "Comments" as a contact list column |
|
67 |
|
| 314 |
Andrew B |
Compiled Applications and Assemblies signed with a certificate so they can be whitelisted in AV and AppLocker. |
|
66 |
|
| 315 |
Wesley |
Add the ability to make custom "call outcomes/disposition" (ie. sale, call back, not interested, DNC)after each call ends/hangup. Add disposition of call result in Call Log. Allow Call Log to be exported to csv or text file. |
|
66 |
|
| 316 |
Juan |
Put a button in the "Answer" dialog, to place the call directly on hold, instead of stopping the ongoing call. |
|
62 |
|
| 317 |
Kevin K |
Please add Fax support for both G.711 and T.38 |
|
62 |
|
| 318 |
Kerstin |
Is it possible to use MicroSIP within Microsoft Teams? Would a Microsoft Teams integrations be useful (https://appsource.microsoft.com/en-US/marketplace/apps?src=wnblogmar2018&product=teams)? |
|
62 |
|
| 319 |
Sven |
Add an automatic updater that downloads and installs the newest version automatically instead of redirecting the user to the website to download it manually. |
|
61 |
|
| 320 |
Leendert-Jan |
The ability to run different commands when different numbers are pressed, making a menu for users. e.g: 'Press 1 to receive the newsletter.'
Not very complex, just different commands for different numbers to press :) |
|
60 |
|
| 321 |
Tekin |
Remote Number (ANI) hide option when call handle. could not see ANI number. |
|
59 |
|
| 322 |
MatthiasSchröder |
Menu option to call mobile number from contact context menu. |
|
59 |
|
| 323 |
Rudolf |
Is there a reason why the slash / is not stripped off like other characters (,.,),- etc..?
The slash is often used in Germany as separator between areacode and number! |
|
58 |
|
| 324 |
Robby |
ability to paste a number (as DTMF signal) within an active call. Lot of business people need this to register the call in to a certain meeting by entering such a meeting ID in Webex or Zoom (234567899#) |
|
57 |
|
| 325 |
George |
Tones optional pack (busy etc). Tones, played by PBX (Asterisk, for example), are not clear. It is better to have internal signals. |
|
57 |
|
| 326 |
Julio |
plantronics headset support |
|
56 |
|
| 327 |
Rafael Tavares |
Add option to disable "show password" on sip account |
|
56 |
|
| 328 |
Kevin K |
Please add option for Late Media Invite. Initial Invite with no SDP and utilize the first Codec returned in the 183/200OK that matches configured Codecs |
|
55 |
|
| 329 |
Astro |
Can't connect to sip.linphone.org (incorrect password error). |
|
54 |
|
| 330 |
Gui |
IMS-AKA authentication support |
|
53 |
|
| 331 |
Alessandro |
Presence (BLF) also on the "Shortcut" Buttons (not only in the personal directory |
|
51 |
|
| 332 |
Danilo |
Support for sending RFC3325 P-asserted-identity. |
|
49 |
|
| 333 |
Panagiotis |
Phonenumber callerID Lookup from a mysql database |
|
49 |
|
| 334 |
Seejay |
Integrate a Hifi audio quality stereo audio codec via OPUS 256kbit/s with inbandFEC enabled. |
|
47 |
|
| 335 |
Astro |
Option for chat history/logging with date/time |
|
47 |
|
| 336 |
kenny |
failover using outbound proxy or dual registration |
|
46 |
|
| 337 |
Greg |
AAC-LD/AAC-ELD Codec Support even if not free. |
|
46 |
|
| 338 |
Johnny |
I wish for the possibility to select two simultanious ring devices as in Skype. The possibility to send/hear the ring signal in both headset and PC speaker for example. |
|
44 |
|
| 339 |
Astro |
sip_auth_clien ...Unsupported digest algorithm "SHA-256" |
|
43 |
|
| 340 |
Oksana |
Contacts have disappeared, it’s very inconvenient, make it possible to save and restore them in cases of such failures. |
|
42 |
|
| 341 |
Jaroslav |
Answer and decline button green backgroud to answer /red to decline :-) |
|
39 |
|
| 342 |
MatthiasSchröder |
Option to auto hide as a sidebar with a small button to unhide. |
|
36 |
|
| 343 |
Jens |
Bei Anruf laufende Audioquelle stummschalten. Frage: wäre es Möglich, eine laufende Audioquelle wie Musik Stream für die dauer des Anrufes stumm zu schalten...
Ansonsten finde ich das Programm super.. weiter so. |
|
35 |
|
| 344 |
John Jairo Zuñiga |
Que me muestren como recuperar registros de llamadas no contestadas o perdidas que fueron eliminadas por el usuario de microsip. |
|
35 |
|
| 345 |
admin |
Ability to lock the account and settings with a password, so that they can't be viewed or changed. Just because you have access to the computer doesn't mean you should have the ability to view the account password or change the configuration. |
|
35 |
|
| 346 |
M2 |
An OPTION to choose a default account, which will be always used after the startup of MicroSIP instead of the last account used before close MicroSIP. |
|
31 |
|
| 347 |
MatthiasSchröder |
Map global Windows hotkey to call the number from active selected windows text. |
|
21 |
|
| 348 |
pp2 |
clear call history "select all" and clear history button required |
|
20 |
|
| 349 |
Malon |
different ringtones, depending on caller
this would be great!
thank you in advance if ever considered worth to implement |
|
19 |
|
| 350 |
Enrique |
Manage Urls when we click on phone numbers with spaces the web-browser adds automatically %20 on spaces separators in numbers like this XXX XXX XXX. So add something to manage that or if come with %20 this part dont put it |
|
15 |
|
| 351 |
Boris |
Special shape of icon in the system tray when DND option is on. (May be, another for AA?) |
|
15 |
|
| 352 |
Rem |
When copy & paste telephone number country code doesn´t follow. +467012345678 shows in the calling list but when we copy and paste it, it will be: 07012345678. Would like to behold the countrycode. |
|
13 |
|
| 353 |
Petrochka |
Add a IAX protocol support. |
|
11 |
|
| 354 |
Tim |
I use the microsip as an easy to use client on a NEC SV9100 system.
nec should also be able to handle presence status.
yet, nec uses port 8383 for presence status.
is there any way I can make microsip more compatible with NEC presence ? |
|
9 |
|
| 355 |
Peter |
Record in multiple channels to separate transmit and receive |
|
9 |
|
| 356 |
Albert |
Subscribe to a contact in the contact list and be notified when this contact is avaiable for calling ( not busy or in a call ) |
|
7 |
|
| 357 |
Eduardo |
Function to generate the password hash from the microsip executable or similar method to generate MicroSIP.ini for remote provisioning.
Running code on the machine:
microsip.exe generate_hash [email protected] |
|
5 |
|
| 358 |
cobwebb |
Strip non numeric characters from numbers passed in via command line (so that %20 and + etc are stripped from poorly formatted callto and tel links) |
|
5 |
|
| 359 |
Zoran |
Web browser plugin - click to call option from web sites. |
|
4 |
|
| 360 |
Bobby |
Don't bring conference window into foreground when dialing a number. After making a call you have to click the dial pad to bring it in foreground to dial more digits on you keyboard. |
|
4 |
|
| 361 |
Bobby |
Add pause ability to dial-plan. A P or comma could be used to insert a pause and allow the call to connect and then dial the remaining digits. Example you dial 2289 it would then dial 16185551212P2289 using a dial plan of xxx In turn would call 618555 |
|
3 |
|
| 362 |
Guy |
Compatibility with Jabra Keypad to mute, hold, anwer |
|
1 |
|
| 363 |
Marco |
DNS SRV resolution for SIP domain |
|
4129 |
 |
| 364 |
JGR |
Multiple SIP accounts |
|
3910 |
|
| 365 |
Sega |
Search within the Contacts (by starting to type the name and the results gradually decrease as one proceeds with typing the name). |
|
3778 |
|
| 366 |
Pietro |
DTMF using SIP INFO method |
|
3110 |
|
| 367 |
Beat We. |
Contacts: Vertical Scrollbar (instead of horizontal bar) |
|
3075 |
|
| 368 |
Beat We. |
Separate contacts.ini or contacts.xml file |
|
2582 |
|
| 369 |
tester |
volumen bar to up/down and Mute the sound of speaker and microphone when we are in conversation |
|
1346 |
|
| 370 |
oondraa |
option to load google contacts |
|
1025 |
|
| 371 |
Chris Cox |
Option to register with Windows as default handler for RFC3966 href="tel:" links. |
|
918 |
|
| 372 |
Pilou42 |
Encrypt password. No password at all should appear in plain text, neither in GUI settings or .ini file. |
|
777 |
|
| 373 |
Dan |
Opus codec support |
|
755 |
|
| 374 |
Rachel |
Direct P2P calls ; IP to IP calls ; In case there is a server outage etc. http://wiki.ekiga.org/index.php/Documentation#Calling_IP_to_IP_directly ; http://wiki.ekiga.org/index.php/Peer_to_Peer |
|
744 |
|
| 375 |
Alex |
Please add support for the Opus Codec. |
|
730 |
|
| 376 |
Iron |
run extern script or aplication on answer and send callerID to the script http://www.microsip.org/help |
|
727 |
|
| 377 |
misus |
accelerated video decoding DXVA for both vga cards Nvidia & ATI |
|
718 |
|
| 378 |
noname |
Support (if installed) for an option to associate MicroSIP with sip: hyper-links - 'Click to Call'. |
|
692 |
|
| 379 |
marduk |
make resizable windows and add minimize, maximize and restore buttons (all windows, chat, video, contacts...), do normal windows please |
|
654 |
|
| 380 |
Giuseppe |
Hold button in Single Call Mode |
|
648 |
|
| 381 |
Simon |
Handle URIs so can click on SIP:xxx links to initiate dialling. |
|
586 |
|
| 382 |
Jonathan Fuentes |
I would love it if you added a Mute Mic button. |
|
578 |
|
| 383 |
Jay |
Ability to use dialpad when in a call (i.e. navigating IVR) |
|
563 |
|
| 384 |
Shaun |
Setting in account to make registration optional (i.e. use only for outbound calls). Some providers only allow 1 client to be connected via SIP. So useful to have hard phone registered for in/out and soft phones on PC's set to out only. For this leave SIP server field empty. |
|
554 |
|
| 385 |
Yann |
Resising of the main window.
The main window cannot be resised
For example when in "calls" menu you need to use the slider whereas there is plenty of space on my PC screen ;)
Alternative solution would be to make it larger so the list in contact would |
|
543 |
|
| 386 |
x-link |
Current call duration counter(Digital clock) is essential. |
|
542 |
|
| 387 |
Denis |
Do not record position and size of the window changes in a .ini file at the time of the change. This gives a great load on the disk and slows down the rendering. Save settings on close/minimize the application. This problem was fixed. Let us know if problem still exists. |
|
542 |
|
| 388 |
Cory |
Call History, including date/time for received, outgoing, and missed calls. |
|
529 |
|
| 389 |
Roberto |
The contact list could show who is busy MicroSIP supports this. Configure your SIP server. |
|
529 |
|
| 390 |
Alex |
--> bigger window
PS: I can't end a call This happend when account configured incorrectly. |
|
526 |
|
| 391 |
salaros |
MicroSIP is not integrated with Windows 7/8 taskbar.
It's not possible to restore a minimized instance by clicking on a pinned MicroSIP icon. When you finally restore it from the tray icon a new non-pinable icon appears (total: 2 icons). |
|
518 |
|
| 392 |
Yann |
Ringing on an other audio device:
When you are away from your desk (like in the other room) you can t hear the ring coming out of your headset. Therefore I have a second audio output I could use to have the ring of the phone.
(xlite does offer such poss |
|
516 |
|
| 393 |
Shafiq |
the close button must minimize to tray.
and minimize button must minimize on task bar. |
|
508 |
|
| 394 |
James |
Add version number (and Publisher) information to the windows installer config. See here: http://img39.imageshack.us/img39/5778/microsip.png |
|
503 |
|
| 395 |
Dmitry |
additional to CmdIncomingCall, add CmdAnswerCall and CmdDisconnectCall http://www.microsip.org/help |
|
502 |
|
| 396 |
bruno |
Store the settings into the profiles of the user, so it become available when using roaming profiles null |
|
494 |
|
| 397 |
LWC |
Speakerphone! That is, a speakerphone button that would use a different sound device than the regular one (i.e. headphones). Just like Nonoh's client has: http://img263.imagevenue.com/img.php?image=23909_nonoh_122_509lo.jpg |
|
477 |
|
| 398 |
Knut |
Please add G.711 μ-law and G.711 A-law. They are preferred by many VoIP providers, and without them the call quality is much worse using G.729 or GSM instead as they don't support the high bandwidth codecs. PCMA and PCMU are the same |
|
468 |
|
| 399 |
bago |
Use default input/output device if the selected one is not available. Back to the selected as soon as it is available. I use USB headset and they are not always connected (as they are on the docking station) and I have to restart MicroSIP each time. |
|
467 |
|
| 400 |
Dallas |
a way to transfer calls |
|
465 |
|
| 401 |
Giles |
X widget - Close to System Tray (does not exit program), to exit program use right click system tray icon. (I often shutdown MicroSIP by mistake, but I want it running all the time). |
|
446 |
|
| 402 |
Knut |
Selected "Ring device" should perhaps apply only to incoming calls? Unnecessary for PC speakers to play sound when I call out with headphones, especially in an office environment. |
|
446 |
|
| 403 |
Guy |
Provide means of clearing the dialled numbers list, everything or individual entries |
|
445 |
|
| 404 |
Alex |
To have the PAUSE button when calling external gateways. Like Nokia phones if you know them...
Cheers, great product!! |
|
440 |
|
| 405 |
Christian |
Ignore incoming calls: Instead of 'declining' calls (which might be considered rude), just ignore them. Mute the ringing and wait until the caller hangs up by himself. |
|
427 |
|
| 406 |
H Br |
Attended call transfer. At present only blind transfer is supported. Thanks! |
|
424 |
|
| 407 |
Paul |
Don't crash a virgin media router when attempting to make a call. Consistent bug with microsip or the modem - but other soft phones work fine :( |
|
409 |
|
| 408 |
Igor |
Lately with cheapvoipinc not working anymore. Possible the timeout is too short. Would be nice if would be adjustable. |
|
388 |
|
| 409 |
WGRichie01 |
first off I love this application the small footprint is great
would love to be able to have 2 extensions from the same PBX connection, I currently provide helpdesk support also and find I have to use 2 applications or manually switch resetting the login |
|
386 |
|
| 410 |
Aliby |
Attended Transfer:
Many hosted SIP protocols will not support Blind Transfer as there is a handover event that most softphones see as an answer on the tranfer line and then drops the call. |
|
385 |
|
| 411 |
Lars |
A option to completly disable the video feature so those video popup windows dont appear everytime a call is recived. |
|
363 |
|
| 412 |
Admin |
video calls |
|
358 |
|
| 413 |
Guillaume |
Option to have the window minimize normally.
I have the program pin to my taskbar with windows 7, but when the window minimize, it goes only to the task icon and clicking on the taskbar starts a second conflicting instance. |
|
356 |
|
| 414 |
YAGrand |
Hello.
When an incoming call, the pop-up window not shows the name and number of the caller. Even if the caller is in the contact list.
Thank you for MicroSip
PS
Russian language. |
|
323 |
|
| 415 |
jaime |
option to transfer the call to another SIP user inside the same Domain |
|
302 |
|
| 416 |
Alex |
I am blind and so I am a screenreader user. When I launch Microsip it starts with the three main tabs, there is no way to get into the window, where I can find the call edit field and the keys for sending DTMF. I'd like to have the ability to klick on the |
|
294 |
|
| 417 |
Nuri |
UTF-8 support in messages. |
|
266 |
|
| 418 |
Greg |
Allow program to handle CALLTO and TEL protocol in browsers. |
|
258 |
|
| 419 |
tester |
option to enable/disable "dump activity log file", in order to save cpu use and disk accesses |
|
246 |
|
| 420 |
Urlich |
Missed calls log |
|
236 |
|
| 421 |
Bago |
Display contact names in the call log tab, for registered contacts, not their numbers. |
|
222 |
|
| 422 |
saravak |
Command switches to start it from a script and call a number! Usage: "microsip.exe number" |
|
206 |
|
| 423 |
JGR |
Option to select audio hardware for input/output |
|
197 |
|
| 424 |
Dink |
Chrome extension/handler for tel: and sip: (Click to call) |
|
194 |
|
| 425 |
RE: Rachel |
Direct P2P calls ; See Ring (was SFLphone); P2P calls by IP or OpenDHT http://ring.cx/en/news#at-the-heart-of-ring-opendht-a-distributed-hash-table |
|
191 |
|
| 426 |
KJ |
Conferencing |
|
170 |
|
| 427 |
tester |
on incoming calls it rings only 1 time, please keep ringing continuously until call is answered |
|
161 |
|
| 428 |
Steve |
Option for enabling / prioritizing voice codec ?
Currently, I am trying to make calls to a polycom phone which supports HD voice. If the Polycom originates the call, 722 is selected. If Microcosip initiates the call, PCM-uLaw is selected. The SDP conta |
|
147 |
|
| 429 |
Vartan |
Separate buttons for 'End' and 'Call'. Currently if you are about to press 'End' and the other party hangs up, your 'End' button becomes 'Call' and you end up ringing them again. |
|
144 |
|
| 430 |
Stu |
Higher bit rates for OPUS Codec. 64kbs or 128kbs would be nice. |
|
142 |
|
| 431 |
Mark |
hold feature |
|
141 |
|
| 432 |
Mark |
transfer call feature |
|
133 |
|
| 433 |
Rus2lan |
Not direct transfer, but allow make call and speak with other number before transfer. Like Number A Hold->Call to B, Speak, hangup-> Number A speak with number B. |
|
133 |
|
| 434 |
Jorg |
I am really looking for an option to do a warm transfer. So i can speak to the party first before i transfer the line. |
|
92 |
|
| 435 |
Michele |
Trying to report an issue but there is no MX record for microsip.org. Thanks! MX are not mandatory. As per RFC2821, if no MX is found the A record will be used. |
|
82 |
|
| 436 |
microsip4ever |
ringing on different audio device and/or ringing on pc speaker.
bloatware x-lite for example can ring on pc speaker under xp. |
|
64 |
|
| 437 |
Youri |
January 2020: the setting for Opus, "maxaveragebitrate=20000" means 8kHz mono bandwidth only, and G.722 sounds better. 50000 would make MicroSIP sound as good as PhonerLite! ;) Thanks! |
|
53 |
|
| 438 |
Luke |
Choose local port instead of "5060" default. Like in version 3.10.5 |
|
48 |
|
| 439 |
TLS support |
Please integrate TLS support like CSipSimple |
|
35 |
|
| 440 |
Admin |
instant messaging |
|
27 |
|
| 441 |
tester |
information about codecs and bandwidth in use, for example: "in:G711-64k _ out:GSM-13k" in a status bar, or title bar |
|
23 |
|
| 442 |
Pieter |
DTMF tones while dialling (for wading though call centre menus) |
|
22 |
|
| 443 |
Knut |
Set bitrate for and fine tune video. I've tested h.264 between 2 microsip clients and compared it with 2 Express Talk clients which uses h.263, and Express Talk's video has far better quality over the same lines. |
|
21 |
|
| 444 |
Afshin |
sip proxy |
|
14 |
|
| 445 |
Pieter |
The ability to edit contacts, rather than just add or delete |
|
12 |
|
| 446 |
Pieter |
It would be nice to be able to choose your own sounds from MP3 or WAV |
|
9 |
|
| 447 |
Steve |
option for putting it into an "AUTO-ANSWER" mode |
|
7 |
|
| 448 |
Knut |
In the current version one must use the mouse to use the dialpad. It would be nice to be able to dial with the numeric keys on the computer keyboard. |
|
4 |
|
| 449 |
JT |
Configuration of outgoing port other than 5060 - or at least use another one when the 5060 is occupied by another application (like the Windows Communicator). |
|
2 |
|
| 450 |
Guyaume B. Parenteau |
Automatically removed dash when we enter phone number :) |
|
2 |
|
| 451 |
Vineet |
call without a regsitar or proxy i.e. only endpoints |
|
1 |
|
| 452 |
Victor |
Choose IP address to bind to (and use in SIP messages). |
|
1 |
|
| 453 |
w2k |
Compatible with previous OS, such as windows2000,
win98. |
|
706 |
 |
| 454 |
RE: Nito |
Please look: """
Able to have multiple plataform windowsmobile, iphone, android, blackberry """
You can search for Zoiper or Linphone too |
|
657 |
|
| 455 |
Nito |
An iOS, WP7.5 - 8, and Android App. |
|
626 |
|
| 456 |
amix |
ActiveX component (scriptable via Windows Script Host). Could place call from other app, could execute programs on local side when called, etc. |
|
510 |
|
| 457 |
noezmeg |
Symbian apps would be great to support a platform independent. |
|
496 |
|
| 458 |
Chris |
SIP SDK for Windows phone 8/8.1 would be greatly appreciated especially if it is simple and easy to use with the same functionality of this program. |
|
334 |
|
| 459 |
Matthew |
Remove the requirement to input domain name. Older version only registered when the domain name was left blank. New version always gets registration timeout (with sip2sip.info) |
|
26 |
|
Downloads
