Asterisk cmd Conference


Simple Conference bridge; alternative to MeetMe that does not require Zaptel/DAHDI.
The third-party addition app_conference is a channel-independent conference application.


Conference is not in the Asterisk standard distribution and can be found at



name is a unique identifier for the conference. It will be created dynamically if it does not exist.

flags is one or more of:

  • M: Moderator (presently same as speaker)
  • S: Speaker (listens too - do not specify both)
  • L: Listener
  • T: "Telephone caller" (just for stats?).
  • V: Do VAD on this caller
  • D: Use Denoise filter on this caller.

Flags for additional patches (See TIP)

  • n: Announce user exit / enter
  • m: Play music on hold if only one person is in the conference (should be 'h' instead of 'm'?!)

priority is not currently used.


Conference is a good alternative to the standard MeetMe because it does not require a timing source to be available. Lack of timing can be a pain when no zaptel hardware is present.


Patch it for more features

Conference does not provide any feedback when users join or leave a conference. There is a patch available for version 2.0.1 which enables music on hold and sound notifications whenever a participant joins or leaves the conference, you can find the download link and more info on how to use it here: appconference moh and nofitications. Also there's an version of the same patch with added operator transfer ability, you can find info aboutit on the same site : appconference operator transfer.

Enter/leave sounds without patching

Here is onother way that doens't require patching / recompiling appconference but unfortunately it's a bit complicated.
In the dialplain:

; This should be the context from which users dial into the bridge
; [...]

; This is the extension that people actually dial to get the bridge
exten => 999999,1,Goto(conf-conferencename,join,1)

; [...]

; Make as many of these contexts as you have seperate conference bridges
; change "conferencename" in each
exten => join,1,System(/opt/asterisk/bin/conference-announce conferencename in)
exten => join,2,Conference(conferencename/S/1)

exten => h,1,System(/opt/asterisk/bin/conference-announce conferencename out)

; make one of these extensions per seperate conference bridge
exten => conf-conferencename,1,Conference(conferencename/S/1)

exten => in,1,Answer()
; if I use Playback here instead of BackGround, asterisk crashes
exten => in,2,BackGround(conf-announce)
exten => in,3,ResponseTimeout(5)
exten => in,4,Hangup()

exten => out,1,Answer()
exten => out,2,BackGround(conf-leave)
exten => out,3,ResponseTimeout(5)
exten => out,4,Hangup()^

In the above dialplan snippet conference-announce is a Perl script that users the Asterisk Manager API to cause the conference to call the "in" or "out" extension in the "confhelper" context which play a sound file that indicates a user arriving or leaving.

Design goals

It has several design goals which are different than meetme:

  • It does not require a zap channel for timing.
  • It is very efficient when used with channels which support DTX (silence detection/discontinuous transmission).
  • It can do VAD on channels which do not support DTX (although this is probably more expensive than just mixing them).
  • It presents messages on the Monitor interface for determine which speakers are active.

I believe that people other than myself are using this, but my environment is pretty rigid, and cases outside of this aren't tested:
  • Some callers come from the PSTN via IAX2 from a box with Zaptel cards, via uLaw.
  • Other callers come from iaxclient based softphones using GSM via IAX2 (with DTX).

Mixing design

  • Minimize encoding/decoding, minimize mixing.
  • Minimize generational loss from trancoding.
  • Usual cases are handled very efficiently:
    • One speaker: That speaker's frame is sent directly to each participant which uses the same codec. It is trancoded _once_ for each additional codec type used by participants.
    • Two speakers: Each speaker gets the other speaker's frames. The two speaker's frames are decoded and mixed, and then encoded _once_ for each codec type used by participants.


Naturally, app_conference is GPL. The CVS also includes parts of libspeex, which is distributed under it's a BSD-style license.

Getting app_conference

app_conference is available from the 'AppConference' project on sourceforge:

Compiling app_conference

For default Asterisk installations, change these lines in Makefile:
INSTALL_PREFIX := /usr (near top of Makefile)

install: all
for x in $(SHAREDOS); do $(INSTALL) -m 755 $$x $(INSTALL_MODULES_DIR) ; done
/usr/sbin/asterisk -rx "restart now" (near bottom of Makefile)

(Note: you will probably also have to make one or more of the chages listed below in the section:
Compile the current cvs version with asterisk 1.0.7)

Using app_conference

There is no configuration file. Conferences are created on-the-fly.

Dialplan syntax:

  • ConferenceName: Whatever you want to name the conference
  • Flags one of more of the following:
    • M: Moderator (presently same as speaker)
    • S: Speaker
    • L: Listener
    • T: "Telephone caller" (just for stats?).
    • V: Do VAD on this caller
    • D: Use Denoise filter on this caller.
  • Priority: Currently ignored; was to be a "speaking priority" so a higher priority caller could "override" others.
  • VADSTART: Optional: "probability" to use to detect start of speech.
  • VADCONTINUE: Optional: "probability" to use to detect continuation of speech.


app_conference doesn't have DTMF-activated features or anything like that.
Doesn't need timers like meetme; I have compiled and now use this under the Xen virtual machine. See the forums at — Giles Morant.
Through the manager interface app_conference will report each user's DTMF digits dialed, and will thus allow you to e.g. programatically play sounds to users etc.


It would be nice to have solid benchmarks to present, but a good size machine should be able to handle many callers when either (a) they are using DTX, or (b) they are listen-only. Probably hundreds.


The iaxclient-devel mailing list is probably as good a place as any to discuss.

Compile the current cvs version with asterisk 1.0.7

(Note: This appears to already be fixed in the released app_conference v2.0.1)

  • modify the makefile to point to the include directory of asterisk header files
  • modify member.c, around line 76-79 comment out the following:

ast_log( AST_CONF_DEBUG, "CHANNEL INFO, CHANNEL => %s, DNID => %s, CALLER_ID => %s, ANI => %s\n",
chan->name, chan->cid.cid_dnid, chan->cid.cid_num, chan->cid.cid_ani ) ;

  • modify conference.c:

add include for asterisk/utils.h at the top of the file
  1. include <asterisk/utils.h>

around line 612 replace pthread_create with ast_pthread_create

  • I also needed to modify cli.c
around line 235 in cli.c
change "ast_copy_string" to "strncpy":

ast_copy_string(newsound->name, file, sizeof(newsound->name));
strncpy(newsound->name, file, sizeof(newsound->name));

app_conference is brought to you by the letter q, and the number e

Setting up conferences

One thing I like about app_meetme is that I can define conferences and passwords. The same can be achieved with app_conference through the database and dialplan for realtime conference creation..

exten => s,1,Wait(1)
exten => s,2,DigitTimeout,5
exten => s,3,ResponseTimeout,8
exten => s,4,BackGround(enter-conf-call-number)
exten => s,5,Waitexten(20)
exten => s,6,Goto(s,4)

exten => _XXXX,1,DBget(pass=conferences/${EXTEN}) ; implied jump to 101 if it doesn't exist
exten => _XXXX,2,GotoIf($["xxx${pass}" = "xxxNONE"]?30) ; if no pass, jump directly to conf
exten => _XXXX,3,Read(secret,pls-enter-conf-password,10)
exten => _XXXX,4,GotoIf($["${pass}" != "${secret}"]?40:30)

exten => _XXXX,30,Conference(${EXTEN}/MTV)
exten => _XXXX,31,Hanugup

exten => _XXXX,40,Playback(conf-invalidpin)
exten => _XXXX,41,Hangup

exten => _XXXX,101,Playback(conf-invalid)
exten => _XXXX,102,Hangup

For example, to create conference 1234 with password 9,

database put conferences 1234 9

for a conference with no password,

database put conferences 1234 NONE

app conference with VICIDIAL

VICIDIAL and app_conference
(1.2 tree) EXPERIMENTAL!!!

app_conference offers the plusses of being able to do conferencing
without any zaptel hardware or timers as well as doing native codec
streaming, meaning no manditory downsampling to slin like meetme.
These two features alone mean a rather significant performance boost
to companies who only use VOIP for agents and trunks as well as not
requiring the cost of zaptel hardware or the nightmare of getting
ztdummy to work on some systems.

As of right now, I have a test system fully up and running using
app_conference with VICIDIAL on Asterisk I have made several
modifications to the app_conference code to make it more of a drop-in
replacement for meetme so that it does not require any changes to the
VICIDIAL codebase. I have added entry and exit sounds(slightly
different from meetme's) as well as "only one in this conference"
message in addition to changing the default member type to speaker(S).
So far I have tested with IAX and SIP channels only and everything
works just like on meetme but I have not done any capacity testing or
tested it in production yet.

If you want to use app_conference instead of meetme for VICIDIAL then follow these instructions:

cd /usr/src/asterisk
cd app_conference
make clean
make install

In your extensions.conf file you would replace these lines:

exten => 8600051,1,Meetme,8600051
exten => 8600052,1,Meetme,8600052
exten => 78600051,1,Meetme,8600051|q
exten => 78600052,1,Meetme,8600052|q
exten => 68600051,1,Meetme,8600051|mq
exten => 68600052,1,Meetme,8600052|mq

with these lines:

exten => 8600051,1,Conference(8600051)
exten => 8600052,1,Conference(8600052)
exten => 78600051,1,Conference(8600051|q)
exten => 78600052,1,Conference(8600052|q)
exten => 68600051,1,Conference(8600051|Lq)
exten => 68600052,1,Conference(8600052|Lq)

Here are the possible feature flags for this version of app_conference:
o M: Moderator (presently same as speaker) [default]
o S: Speaker
o L: Listener
o T: "Telephone caller" (just for stats?).
o V: Do VAD on this caller
o D: Use Denoise filter on this caller.
o d: Send manager events when DTMF is received.
o q: Do not play enter or exit sounds.
o i: use inband dtmf broadcast.
o t: use rfc dtmf signal broadcast.

See also

Created by: vandry, Last modification: Sun 06 of May, 2012 (17:39) by admin
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