Asterisk Wishlist

Asterisk Wishlist

Remember: you can ask for features as long as you want. But that won't make such features appear magically. Basically, you have these options:
  • "get dirty", learn how to program and submit code to Asterisk
  • convince another programmer that your wanted feature is valuable to them so that this programmers does it for you (and the rest of the community)
  • Open your pocket and put some bounty onto a feature and wait until someone wants to earn some money. Hey, even programmers have to eat :-)


  • SecurityManager: ability to set a custom number of cyphers to check (now it is fixed to 10). E.G. letting the anonymous caller dial in his/her number and finish with a pound sign, so that any number - short, local or international - can be accepted by the SecurityManager (Mark)
  • Clustering: Add support for an Active-Active cluster that even could do load balancing. Cluster should be smart enought to control not only Asterisk crashes, but also be able to redirect service to the working node when Asterisk is hang (or not working correctly). For now, consider High Availability Asterisk (HAAst)
  • Nagios Integration: Integrating Asterisk with nagios (see http:\\ ), so we can check always that Asterisk is working correclty
  • SER Integration: Either Asterisk should be able to remplace easily its own SIP stack with the one that comes with SER, or Asterisk should have the feautures that SER have (i.e, SIP Proxy, etc)
  • LDAP integration: LDAP should be better integrated on Asterisk. Most companies are using LDAP as their Corporate Directory. So, this would be a real and fast way to integrate Asterisk with Corporate Directories.
  • SRTP Support: Be able to encrypt calls - partly implemented, see
  • Add support to resolve IP of person calling and base dialplan actions according to there country. Example would be for 911 support, when being used out of country would play a wav instead of calling the call centre. - BenJP2k1
  • Add CSTA compliance to allow Asterisk to work with CTI Standards Applications ( Presence, Altitude or Genesys for Example).
  • Make analog FAX machines work reliably (and fast) behind Asterisk
  • Dialplan conditionals, DoIF and IF in dialplan - JKWoss
  • Implement CNG, VAD and DTX - MustDie
  • Logging option for asterisk to continuously write a console log that mirrors EXACTLY what you see on the console
  • take the service codes like *70, *69 out of chan_zap and do it in a generic, configurable way so that we can have an while-connected-dialplan that listens to DTMF and can make things happen accordingly.
  • IPv6 Support for SIP/IAX2/H323 and Manager API
  • Implement a new jitterbuffer for asterisk, supporting multiple channels, DTX, PLC, and other TLAs. (now in asterisk-CVS)
  • Support multi-prompt playback (list/array/etc. of prompt file names) (check bug 3337 - Moc)
  • Support full Silence Suppression on RTP, effictively reducing bandwidth consumption by 50% or so.
  • Support tuning the RTP sample period from today's hardcoded 20ms as to allow for low-bandwidth VoIP connections trading off some latency.
  • Add a way to configure a outbound route different for each extention.
  • Wideband Codec support
  • Call quality information available to dial plan. This information should really be sent both directions through IAX2 so we can know if we are being heard well not only if we can hear. Information of use would be jitter, packet loss, latency, etc. If a call quality is suffering via one route we need to dynamically be able to reroute calls.

Command line interface

  • a means to mask out passwords in the console output and make it the default - benjk
  • a means to set filters for debug output filtering by address and by event type - benjk
  • make reload work for ALL the configuration.. Meetme.conf/zapata.conf.... - Moc


  • Add ability to suppress extension number announcement in Voicemail answer message: e.g. "The person at the number you have dialed is not available at the moment; please leave ..." rather than "The person at the extension XYZ is not...". (Mark)
  • Create a new system to handle new voicemail and lockfile - Moc
  • Be able to cancel a voicemail - Moc
  • Dont show MWI until the voicemail has being finish and 'accepted' - Moc
  • Be able to send a voicemail to a user - Moc
  • Be able to call user external number if voicemail is in their mailbox (Like a octel system, can call multiple person 1 person at the time) - Moc
  • Make options like numbers and setting configurable. ariel_
  • Host mail app on one server and SIP registration on another. Pass the MWI messages as approprate. (perhaps mailbox=1234@anotherhost|voicemailContect)
  • Make play sound configurable - pixies



  • New feature to be put already in the future app_conference: bugs 1562 - Moc
  • Better configuration file see Example proposed configuration file - Moc
  • Multiple layer of users, maybe 1 PIN per user predefined in a database somewhere - Moc
  • Have the posibility for the chairman to record the conference - Moc
  • Better management interface, maybe via the Manager API - Moc
  • Video conference support (ISDN Maybe ?) (Split on 4 windows, detect speaker...) - Moc
  • Better streaming facility - Dorian
  • See if a user does echo into the conference - Moc
  • Be able to listen to a specific channel (monitoring/debuging/testing usage) - Moc
  • (Along the same lines - Round-Robbin Muting + Channel ID, for weeding out funky channels/annoyances. - twisted) @Work (Fabe)
  • Ability to ban callers based on either A) Personalized PINS, or B) IP Address. - twisted
  • Having changable GSM/wav.. enter and leave tone
  • Add ability to play a gsm file to all conferences, all users in a specific conference, or a specific user in a specific conference - E|nyPRI
  • Add ability to saynumber the number of people in the conferences (ie: *9) - E|nyPRI @Work (Fabe)
  • Add ability to saynumber the current conference number you are in (ie: *8) - E|nyPRI @Work (Fabe)
  • Add ability to exit on ANY touchtone, not just # (or provide a list of keys that exit) - E|nyPRI - Could be done with 'X' which exits into the pressed extension

Zapata - X100P/FXO

  • Listen for dial tone before dialing - Syncros
  • Define context based on dial tone (ex: message waiting in telco voicemail system) - Syncros
  • Support for Japanese CLID - benjk (on behalf of mack_jpn, isamar and self) | Specification in English (PDF)
  • Have a tool to calibrate the audio level on all FXO/FXS - Moc
  • Support Call Progress Detection on French network with "callprogress" option set to "yes"

Zapata - T1/E1

  • Somehow add a ${PRI_PRIVACY} variable to the channel if the PRI has the callerid privacy flag set, so we can mask/hide callerid on redirected calls - E|nyPRI
  • Ability to control the dialtone on a per channel basis (e.g. offer stutter dialtone if voicemail waiting). I think you'll find it's only on FXS lines, I need it for legacy PBX integration over PRI where I'd choose the stutter in the dialplan
  • Listen for dial tone before dialing


  • Get a t.38 fax driver - ariel_
  • have Spandsp as part of asterisk- ariel_
  • documentation on how to integrate Asterisk with HylaFax


  • Encryption - Conference Call
  • Passthrought MWI messages between system (For seperating Voicemail from the main System) - Moc
  • Have resolve the DNS for the INFO tag for the IAX server IP - bkw_
  • provide an Unregister() command or method
  • get callgroups working for IAX clients


  • Have a register=yes option within a [context] in sip.conf. instead of doing register=>.... before any context - bkw_
  • provide an Unregister() command or method
  • Support for call pick-up by pressing a blinking (!) line buttton on SNOM and similar phones during "ringing" state
  • Encryption
  • fix the type=... mess for incoming calls (currently type has no effect, and that _last_ matching sip.conf entry applies)
  • trigger an event when client registers so the action can be taken in that moment, add something like "onregister=context/extension/priority"


  • Make H323 pre-configured as part of asterisk just like sip and iax.

Manager API

  • Md5 password hashing for Login action. - MustDie
  • Incorporate "Application" field into every "Newexten" event. - MustDie
  • Option to hide/blackout the CallerID fields on events if needed to protect privacy - E|nyPRI
  • Include a standard middleware with Asterisk to circumvent the connect/ disconnect problems with the manager API

Build Process / Runtime Issues (Damin 7/14/2004)

  • Discussion Point: Should asterisk continue to run as root or under its own userid?
  • Discussion Point: Should the Makefiles include "uninstall" options?
  • Discussion Point: Should the Makefiles include Distribution Specific Scripts and Build logic? I.E. "make install debian"

ADSI phones

  • Add the ability to have multiple line appearances on a single line adsi phone using the 6 soft buttons.


  • Need support for a more recent version of Festival before the old ones vanish for download


  • Support for (VoiceXML) directly trough application for develop hi-hend IVR systems

PLEASE add more...


Created by: mochouinard, Last modification: Thu 12 of Jun, 2014 (03:01 UTC) by ocgltd
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