@ -40,6 +40,464 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
# include "asterisk/stringfields.h"
# include "asterisk/pbx.h"
/*** DOCUMENTATION
< configInfo name = " app_confbridge " language = " en_US " >
< synopsis > Conference Bridge Application < / synopsis >
< configFile name = " confbridge.conf " >
< configObject name = " global " >
< synopsis > Unused , but reserved . < / synopsis >
< / configObject >
< configObject name = " user_profile " >
< synopsis > A named profile to apply to specific callers . < / synopsis >
< description > < para > Callers in a ConfBridge have a profile associated with them
that determine their options . A configuration section is determined to be a
user_profile when the < literal > type < / literal > parameter has a value
of < literal > user < / literal > .
< / para > < / description >
< configOption name = " type " >
< synopsis > Define this configuration category as a user profile . < / synopsis >
< description > < para > The type parameter determines how a context in the
configuration file is interpreted . < / para >
< enumlist >
< enum name = " user " > < para > Configure the context as a < replaceable > user_profile < / replaceable > < / para > < / enum >
< enum name = " bridge " > < para > Configure the context as a < replaceable > bridge_profile < / replaceable > < / para > < / enum >
< enum name = " menu " > < para > Configure the context as a < replaceable > menu < / replaceable > < / para > < / enum >
< / enumlist >
< / description >
< / configOption >
< configOption name = " admin " >
< synopsis > Sets if the user is an admin or not < / synopsis >
< / configOption >
< configOption name = " marked " >
< synopsis > Sets if this is a marked user or not < / synopsis >
< / configOption >
< configOption name = " startmuted " >
< synopsis > Sets if all users should start out muted < / synopsis >
< / configOption >
< configOption name = " music_on_hold_when_empty " >
< synopsis > Play MOH when user is alone or waiting on a marked user < / synopsis >
< / configOption >
< configOption name = " quiet " >
< synopsis > Silence enter / leave prompts and user intros for this user < / synopsis >
< / configOption >
< configOption name = " announce_user_count " >
< synopsis > Sets if the number of users should be announced to the user < / synopsis >
< / configOption >
< configOption name = " announce_user_count_all " >
< synopsis > Announce user count to all the other users when this user joins < / synopsis >
< description > < para > Sets if the number of users should be announced to all the other users
in the conference when this user joins . This option can be either set to ' yes ' or
a number . When set to a number , the announcement will only occur once the user
count is above the specified number .
< / para > < / description >
< / configOption >
< configOption name = " announce_only_user " >
< synopsis > Announce to a user when they join an empty conference < / synopsis >
< / configOption >
< configOption name = " wait_marked " >
< synopsis > Sets if the user must wait for a marked user to enter before joining a conference < / synopsis >
< / configOption >
< configOption name = " end_marked " >
< synopsis > Kick the user from the conference when the last marked user leaves < / synopsis >
< / configOption >
< configOption name = " talk_detection_events " >
< synopsis > Set whether or not notifications of when a user begins and ends talking should be sent out as events over AMI < / synopsis >
< / configOption >
< configOption name = " dtmf_passthrough " >
< synopsis > Sets whether or not DTMF should pass through the conference < / synopsis >
< / configOption >
< configOption name = " announce_join_leave " >
< synopsis > Prompt user for their name when joining a conference and play it to the conference when they enter < / synopsis >
< / configOption >
< configOption name = " pin " >
< synopsis > Sets a PIN the user must enter before joining the conference < / synopsis >
< / configOption >
< configOption name = " music_on_hold_class " >
< synopsis > The MOH class to use for this user < / synopsis >
< / configOption >
< configOption name = " announcement " >
< synopsis > Sound file to play to the user when they join a conference < / synopsis >
< / configOption >
< configOption name = " denoise " >
< synopsis > Apply a denoise filter to the audio before mixing < / synopsis >
< description > < para > Sets whether or not a denoise filter should be applied
to the audio before mixing or not . Off by default . Requires
codec_speex to be built and installed . Do not confuse this option
with drop_silence . Denoise is useful if there is a lot of background
noise for a user as it attempts to remove the noise while preserving
the speech . This option does NOT remove silence from being mixed into
the conference and does come at the cost of a slight performance hit .
< / para > < / description >
< / configOption >
< configOption name = " dsp_drop_silence " >
< synopsis > Drop what Asterisk detects as silence from audio sent to the bridge < / synopsis >
< description > < para >
This option drops what Asterisk detects as silence from
entering into the bridge . Enabling this option will drastically
improve performance and help remove the buildup of background
noise from the conference . Highly recommended for large conferences
due to its performance enhancements .
< / para > < / description >
< / configOption >
< configOption name = " dsp_silence_threshold " >
< synopsis > The number of milliseconds of detected silence necessary to trigger silence detection < / synopsis >
< description > < para >
The time in milliseconds of sound falling within the what
the dsp has established as baseline silence before a user
is considered be silent . This value affects several
operations and should not be changed unless the impact
on call quality is fully understood . < / para >
< para > What this value affects internally : < / para >
< para >
1. When talk detection AMI events are enabled , this value
determines when the user has stopped talking after a
period of talking . If this value is set too low
AMI events indicating the user has stopped talking
may get falsely sent out when the user briefly pauses
during mid sentence .
< / para >
< para >
2. The drop_silence option depends on this value to
determine when the user ' s audio should begin to be
dropped from the conference bridge after the user
stops talking . If this value is set too low the user ' s
audio stream may sound choppy to the other participants .
This is caused by the user transitioning constantly from
silence to talking during mid sentence .
< / para >
< para >
The best way to approach this option is to set it slightly above
the maximum amount of ms of silence a user may generate during
natural speech .
< / para >
< para > By default this value is 2500 ms . Valid values are 1 through 2 ^ 31. < / para >
< / description >
< / configOption >
< configOption name = " dsp_talking_threshold " >
< synopsis > The number of milliseconds of detected non - silence necessary to triger talk detection < / synopsis >
< description > < para >
The time in milliseconds of sound above what the dsp has
established as base line silence for a user before a user
is considered to be talking . This value affects several
operations and should not be changed unless the impact on
call quality is fully understood . < / para >
< para >
What this value affects internally :
< / para >
< para >
1. Audio is only mixed out of a user ' s incoming audio stream
if talking is detected . If this value is set too
loose the user will hear themselves briefly each
time they begin talking until the dsp has time to
establish that they are in fact talking .
< / para >
< para >
2. When talk detection AMI events are enabled , this value
determines when talking has begun which results in
an AMI event to fire . If this value is set too tight
AMI events may be falsely triggered by variants in
room noise .
< / para >
< para >
3. The drop_silence option depends on this value to determine
when the user ' s audio should be mixed into the bridge
after periods of silence . If this value is too loose
the beginning of a user ' s speech will get cut off as they
transition from silence to talking .
< / para >
< para > By default this value is 160 ms . Valid values are 1 through 2 ^ 31 < / para >
< / description >
< / configOption >
< configOption name = " jitterbuffer " >
< synopsis > Place a jitter buffer on the user ' s audio stream before audio mixing is performed < / synopsis >
< description > < para >
Enabling this option places a jitterbuffer on the user ' s audio stream
before audio mixing is performed . This is highly recommended but will
add a slight delay to the audio . This option is using the < literal > JITTERBUFFER < / literal >
dialplan function ' s default adaptive jitterbuffer . For a more fine tuned
jitterbuffer , disable this option and use the < literal > JITTERBUFFER < / literal > dialplan function
on the user before entering the ConfBridge application .
< / para > < / description >
< / configOption >
< configOption name = " template " >
< synopsis > When using the CONFBRIDGE dialplan function , use a user profile as a template for creating a new temporary profile < / synopsis >
< / configOption >
< / configObject >
< configObject name = " bridge_profile " >
< synopsis > A named profile to apply to specific bridges . < / synopsis >
< description > < para > ConfBridge bridges have a profile associated with them
that determine their options . A configuration section is determined to be a
< literal > bridge_profile < / literal > when the < literal > type < / literal > parameter has a value
of < literal > bridge < / literal > .
< / para > < / description >
< configOption name = " type " >
< synopsis > Define this configuration category as a bridge profile < / synopsis >
< description > < para > The type parameter determines how a context in the
configuration file is interpreted . < / para >
< enumlist >
< enum name = " user " > < para > Configure the context as a < replaceable > user_profile < / replaceable > < / para > < / enum >
< enum name = " bridge " > < para > Configure the context as a < replaceable > bridge_profile < / replaceable > < / para > < / enum >
< enum name = " menu " > < para > Configure the context as a < replaceable > menu < / replaceable > < / para > < / enum >
< / enumlist >
< / description >
< / configOption >
< configOption name = " jitterbuffer " >
< synopsis > Place a jitter buffer on the conference ' s audio stream < / synopsis >
< / configOption >
< configOption name = " internal_sample_rate " >
< synopsis > Set the internal native sample rate for mixing the conference < / synopsis >
< description > < para >
Sets the internal native sample rate the
conference is mixed at . This is set to automatically
adjust the sample rate to the best quality by default .
Other values can be anything from 8000 - 192000. If a
sample rate is set that Asterisk does not support , the
closest sample rate Asterisk does support to the one requested
will be used .
< / para > < / description >
< / configOption >
< configOption name = " mixing_interval " >
< synopsis > Sets the internal mixing interval in milliseconds for the bridge < / synopsis >
< description > < para >
Sets the internal mixing interval in milliseconds for the bridge . This
number reflects how tight or loose the mixing will be for the conference .
In order to improve performance a larger mixing interval such as 40 ms may
be chosen . Using a larger mixing interval comes at the cost of introducing
larger amounts of delay into the bridge . Valid values here are 10 , 20 , 40 ,
or 80.
< / para > < / description >
< / configOption >
< configOption name = " record_conference " >
< synopsis > Record the conference starting with the first active user ' s entrance and ending with the last active user ' s exit < / synopsis >
< description > < para >
Records the conference call starting when the first user
enters the room , and ending when the last user exits the room .
The default recorded filename is
< filename > ' confbridge - $ { name of conference bridge } - $ { start time } . wav < / filename >
and the default format is 8 khz slinear . This file will be
located in the configured monitoring directory in asterisk . conf .
< / para > < / description >
< / configOption >
< configOption name = " record_file " default = " confbridge-${name of conference bridge}-${start time}.wav " >
< synopsis > The filename of the conference recording < / synopsis >
< description > < para >
When record_conference is set to yes , the specific name of the
record file can be set using this option . Note that since multiple
conferences may use the same bridge profile , this may cause issues
depending on the configuration . It is recommended to only use this
option dynamically with the < literal > CONFBRIDGE ( ) < / literal > dialplan function . This
allows the record name to be specified and a unique name to be chosen .
By default , the record_file is stored in Asterisk ' s spool / monitor directory
with a unique filename starting with the ' confbridge ' prefix .
< / para > < / description >
< / configOption >
< configOption name = " video_mode " >
< synopsis > Sets how confbridge handles video distribution to the conference participants < / synopsis >
< description > < para >
Sets how confbridge handles video distribution to the conference participants .
Note that participants wanting to view and be the source of a video feed
_MUST_ be sharing the same video codec . Also , using video in conjunction with
with the jitterbuffer currently results in the audio being slightly out of sync
with the video . This is a result of the jitterbuffer only working on the audio
stream . It is recommended to disable the jitterbuffer when video is used . < / para >
< enumlist >
< enum name = " none " >
< para > No video sources are set by default in the conference . It is still
possible for a user to be set as a video source via AMI or DTMF action
at any time . < / para >
< / enum >
< enum name = " follow_talker " >
< para > The video feed will follow whoever is talking and providing video . < / para >
< / enum >
< enum name = " last_marked " >
< para > The last marked user to join the conference with video capabilities
will be the single source of video distributed to all participants .
If multiple marked users are capable of video , the last one to join
is always the source , when that user leaves it goes to the one who
joined before them . < / para >
< / enum >
< enum name = " first_marked " >
< para > The first marked user to join the conference with video capabilities
is the single source of video distribution among all participants . If
that user leaves , the marked user to join after them becomes the source . < / para >
< / enum >
< / enumlist >
< / description >
< / configOption >
< configOption name = " max_members " >
< synopsis > Limit the maximum number of participants for a single conference < / synopsis >
< description > < para >
This option limits the number of participants for a single
conference to a specific number . By default conferences
have no participant limit . After the limit is reached , the
conference will be locked until someone leaves . Note however
that an Admin user will always be alowed to join the conference
regardless if this limit is reached or not .
< / para > < / description >
< / configOption >
< configOption name = " ^sound_ " >
< synopsis > Override the various conference bridge sound files < / synopsis >
< description > < para >
All sounds in the conference are customizable using the bridge profile options below .
Simply state the option followed by the filename or full path of the filename after
the option . Example : < literal > sound_had_joined = conf - hasjoin < / literal > This will play the < literal > conf - hasjoin < / literal >
sound file found in the sounds directory when announcing someone ' s name is joining the
conference . < / para >
< enumlist >
< enum name = " sound_join " > < para > The sound played to everyone when someone enters the conference . < / para > < / enum >
< enum name = " sound_leave " > < para > The sound played to everyone when someone leaves the conference . < / para > < / enum >
< enum name = " sound_has_joined " > < para > The sound played before announcing someone ' s name has
joined the conference . This is used for user intros .
Example < literal > " _____ has joined the conference " < / literal > < / para > < / enum >
< enum name = " sound_has_left " > < para > The sound played when announcing someone ' s name has
left the conference . This is used for user intros .
Example < literal > " _____ has left the conference " < / literal > < / para > < / enum >
< enum name = " sound_kicked " > < para > The sound played to a user who has been kicked from the conference . < / para > < / enum >
< enum name = " sound_muted " > < para > The sound played when the mute option it toggled on . < / para > < / enum >
< enum name = " sound_unmuted " > < para > The sound played when the mute option it toggled off . < / para > < / enum >
< enum name = " sound_only_person " > < para > The sound played when the user is the only person in the conference . < / para > < / enum >
< enum name = " sound_only_one " > < para > The sound played to a user when there is only one other
person is in the conference . < / para > < / enum >
< enum name = " sound_there_are " > < para > The sound played when announcing how many users there
are in a conference . < / para > < / enum >
< enum name = " sound_other_in_party " > < para > This file is used in conjunction with < literal > sound_there_are < / literal >
when announcing how many users there are in the conference .
The sounds are stringed together like this .
< literal > " sound_there_are " $ { number of participants } " sound_other_in_party " < / literal > < / para > < / enum >
< enum name = " sound_place_into_conference " > < para > The sound played when someone is placed into the conference
after waiting for a marked user . < / para > < / enum >
< enum name = " sound_wait_for_leader " > < para > The sound played when a user is placed into a conference that
can not start until a marked user enters . < / para > < / enum >
< enum name = " sound_leader_has_left " > < para > The sound played when the last marked user leaves the conference . < / para > < / enum >
< enum name = " sound_get_pin " > < para > The sound played when prompting for a conference pin number . < / para > < / enum >
< enum name = " sound_invalid_pin " > < para > The sound played when an invalid pin is entered too many times . < / para > < / enum >
< enum name = " sound_locked " > < para > The sound played to a user trying to join a locked conference . < / para > < / enum >
< enum name = " sound_locked_now " > < para > The sound played to an admin after toggling the conference to locked mode . < / para > < / enum >
< enum name = " sound_unlocked_now " > < para > The sound played to an admin after toggling the conference to unlocked mode . < / para > < / enum >
< enum name = " sound_error_menu " > < para > The sound played when an invalid menu option is entered . < / para > < / enum >
< / enumlist >
< / description >
< / configOption >
< configOption name = " template " >
< synopsis > When using the CONFBRIDGE dialplan function , use a bridge profile as a template for creating a new temporary profile < / synopsis >
< / configOption >
< / configObject >
< configObject name = " menu " >
< synopsis > A conference user menu < / synopsis >
< description >
< para > Conference users , as defined by a < literal > conf_user < / literal > ,
can have a DTMF menu assigned to their profile when they enter the
< literal > ConfBridge < / literal > application . < / para >
< / description >
< configOption name = " type " >
< synopsis > Define this configuration category as a menu < / synopsis >
< description > < para > The type parameter determines how a context in the
configuration file is interpreted . < / para >
< enumlist >
< enum name = " user " > < para > Configure the context as a < replaceable > user_profile < / replaceable > < / para > < / enum >
< enum name = " bridge " > < para > Configure the context as a < replaceable > bridge_profile < / replaceable > < / para > < / enum >
< enum name = " menu " > < para > Configure the context as a < replaceable > menu < / replaceable > < / para > < / enum >
< / enumlist >
< / description >
< / configOption >
< configOption name = " ^[0-9A-D*#]+$ " >
< synopsis > DTMF sequences to assign various confbridge actions to < / synopsis >
< description > < para > - - - ConfBridge Menu Options - - - < / para >
< para > The ConfBridge application also has the ability to apply custom DTMF menus to
each channel using the application . Like the User and Bridge profiles a menu
is passed in to ConfBridge as an argument in the dialplan . < / para >
< para > Below is a list of menu actions that can be assigned to a DTMF sequence . < / para >
< note > < para >
A single DTMF sequence can have multiple actions associated with it . This is
accomplished by stringing the actions together and using a < literal > , < / literal > as the
delimiter . Example : Both listening and talking volume is reset when < literal > 5 < / literal > is
pressed . < literal > 5 = reset_talking_volume , reset_listening_volume < / literal > < / para > < / note >
< enumlist >
< enum name = " playback(filename&filename2&...) " > < para >
< literal > playback < / literal > will play back an audio file to a channel
and then immediately return to the conference .
This file can not be interupted by DTMF .
Multiple files can be chained together using the
< literal > & amp ; < / literal > character . < / para > < / enum >
< enum name = " playback_and_continue(filename&filename2&...) " > < para >
< literal > playback_and_continue < / literal > will
play back a prompt while continuing to
collect the dtmf sequence . This is useful
when using a menu prompt that describes all
the menu options . Note however that any DTMF
during this action will terminate the prompts
playback . Prompt files can be chained together
using the < literal > & amp ; < / literal > character as a delimiter . < / para > < / enum >
< enum name = " toggle_mute " > < para >
Toggle turning on and off mute . Mute will make the user silent
to everyone else , but the user will still be able to listen in .
continue to collect the dtmf sequence . < / para > < / enum >
< enum name = " no_op " > < para >
This action does nothing ( No Operation ) . Its only real purpose exists for
being able to reserve a sequence in the config as a menu exit sequence . < / para > < / enum >
< enum name = " decrease_listening_volume " > < para >
Decreases the channel ' s listening volume . < / para > < / enum >
< enum name = " increase_listening_volume " > < para >
Increases the channel ' s listening volume . < / para > < / enum >
< enum name = " reset_listening_volume " > < para >
Reset channel ' s listening volume to default level . < / para > < / enum >
< enum name = " decrease_talking_volume " > < para >
Decreases the channel ' s talking volume . < / para > < / enum >
< enum name = " increase_talking_volume " > < para >
Increases the channel ' s talking volume . < / para > < / enum >
< enum name = " reset_talking_volume " > < para >
Reset channel ' s talking volume to default level . < / para > < / enum >
< enum name = " dialplan_exec(context,exten,priority) " > < para >
The < literal > dialplan_exec < / literal > action allows a user
to escape from the conference and execute
commands in the dialplan . Once the dialplan
exits the user will be put back into the
conference . The possibilities are endless ! < / para > < / enum >
< enum name = " leave_conference " > < para >
This action allows a user to exit the conference and continue
execution in the dialplan . < / para > < / enum >
< enum name = " admin_kick_last " > < para >
This action allows an Admin to kick the last participant from the
conference . This action will only work for admins which allows
a single menu to be used for both users and admins . < / para > < / enum >
< enum name = " admin_toggle_conference_lock " > < para >
This action allows an Admin to toggle locking and
unlocking the conference . Non admins can not use
this action even if it is in their menu . < / para > < / enum >
< enum name = " set_as_single_video_src " > < para >
This action allows any user to set themselves as the
single video source distributed to all participants .
This will make the video feed stick to them regardless
of what the < literal > video_mode < / literal > is set to . < / para > < / enum >
< enum name = " release_as_single_video_src " > < para >
This action allows a user to release themselves as
the video source . If < literal > video_mode < / literal > is not set to < literal > none < / literal >
this action will result in the conference returning to
whatever video mode the bridge profile is using . < / para >
< para > Note that this action will have no effect if the user
is not currently the video source . Also , the user is
not guaranteed by using this action that they will not
become the video source again . The bridge will return
to whatever operation the < literal > video_mode < / literal > option is set to
upon release of the video src . < / para > < / enum >
< enum name = " admin_toggle_mute_participants " > < para >
This action allows an administrator to toggle the mute
state for all non - admins within a conference . All
admin users are unaffected by this option . Note that all
users , regardless of their admin status , are notified
that the conference is muted . < / para > < / enum >
< enum name = " participant_count " > < para >
This action plays back the number of participants currently
in a conference < / para > < / enum >
< / enumlist >
< / description >
< / configOption >
< / configObject >
< / configFile >
< / configInfo >
* * */
struct confbridge_cfg {
struct ao2_container * bridge_profiles ;
struct ao2_container * user_profiles ;
@ -81,6 +539,7 @@ static void *bridge_profile_find(struct ao2_container *container, const char *ca
static struct aco_type bridge_type = {
. type = ACO_ITEM ,
. name = " bridge_profile " ,
. category_match = ACO_BLACKLIST ,
. category = " ^general$ " ,
. matchfield = " type " ,
@ -117,6 +576,7 @@ static void *user_profile_find(struct ao2_container *container, const char *cate
static struct aco_type user_type = {
. type = ACO_ITEM ,
. name = " user_profile " ,
. category_match = ACO_BLACKLIST ,
. category = " ^general$ " ,
. matchfield = " type " ,
@ -147,6 +607,7 @@ static void *menu_find(struct ao2_container *container, const char *category)
static struct aco_type menu_type = {
. type = ACO_ITEM ,
. name = " menu " ,
. category_match = ACO_BLACKLIST ,
. category = " ^general$ " ,
. matchfield = " type " ,
@ -164,6 +625,7 @@ static struct aco_type *user_types[] = ACO_TYPES(&user_type);
/* The general category is reserved, but unused */
static struct aco_type general_type = {
. type = ACO_GLOBAL ,
. name = " global " ,
. category_match = ACO_WHITELIST ,
. category = " ^general$ " ,
} ;
@ -1293,8 +1755,6 @@ int conf_load_config(int reload)
/* User options */
aco_option_register ( & cfg_info , " type " , ACO_EXACT , user_types , NULL , OPT_NOOP_T , 0 , 0 ) ;
aco_option_register ( & cfg_info , " type " , ACO_EXACT , bridge_types , NULL , OPT_NOOP_T , 0 , 0 ) ;
aco_option_register ( & cfg_info , " type " , ACO_EXACT , menu_types , NULL , OPT_NOOP_T , 0 , 0 ) ;
aco_option_register ( & cfg_info , " admin " , ACO_EXACT , user_types , " no " , OPT_BOOLFLAG_T , 1 , FLDSET ( struct user_profile , flags ) , USER_OPT_ADMIN ) ;
aco_option_register ( & cfg_info , " marked " , ACO_EXACT , user_types , " no " , OPT_BOOLFLAG_T , 1 , FLDSET ( struct user_profile , flags ) , USER_OPT_MARKEDUSER ) ;
aco_option_register ( & cfg_info , " startmuted " , ACO_EXACT , user_types , " no " , OPT_BOOLFLAG_T , 1 , FLDSET ( struct user_profile , flags ) , USER_OPT_STARTMUTED ) ;
@ -1321,6 +1781,7 @@ int conf_load_config(int reload)
aco_option_register_custom ( & cfg_info , " template " , ACO_EXACT , user_types , NULL , user_template_handler , 0 ) ;
/* Bridge options */
aco_option_register ( & cfg_info , " type " , ACO_EXACT , bridge_types , NULL , OPT_NOOP_T , 0 , 0 ) ;
aco_option_register ( & cfg_info , " jitterbuffer " , ACO_EXACT , bridge_types , " no " , OPT_BOOLFLAG_T , 1 , FLDSET ( struct bridge_profile , flags ) , USER_OPT_JITTERBUFFER ) ;
/* "auto" will fail to parse as a uint, but we use PARSE_DEFAULT to set the value to 0 in that case, which is the value that auto resolves to */
aco_option_register ( & cfg_info , " internal_sample_rate " , ACO_EXACT , bridge_types , " 0 " , OPT_UINT_T , PARSE_DEFAULT , FLDSET ( struct bridge_profile , internal_sample_rate ) , 0 ) ;
@ -1334,6 +1795,7 @@ int conf_load_config(int reload)
aco_option_register_custom ( & cfg_info , " template " , ACO_EXACT , bridge_types , NULL , bridge_template_handler , 0 ) ;
/* Menu options */
aco_option_register ( & cfg_info , " type " , ACO_EXACT , menu_types , NULL , OPT_NOOP_T , 0 , 0 ) ;
aco_option_register_custom ( & cfg_info , " ^[0-9A-D*#]+$ " , ACO_REGEX , menu_types , NULL , menu_option_handler , 0 ) ;
if ( aco_process_config ( & cfg_info , reload ) = = ACO_PROCESS_ERROR ) {
Matthew Jordan
12 years ago