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voice-client: Added default config files to not get erorrs from Asterisk modules.

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ref: #16273
This commit is contained in:
Kent Ekholm 2019-01-16 14:10:09 +01:00
parent 8353c17ea6
commit 57bd744aba
6 changed files with 628 additions and 0 deletions
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80
voice-client/files/etc/asterisk_templates/acl.conf.TEMPLATE Normal file
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;
; Named Access Control Lists (ACLs)
;
; A convenient way to share acl definitions
;
; This configuration file is read on startup
;
; CLI Commands
; -----------------------------------------------------------
; acl show Show all named ACLs configured
; acl show <name> Show contents of a particular named ACL
; reload acl Reload configuration file
;
; Any configuration that uses ACLs which has been made to be able to use named
; ACLs will specify a named ACL with the 'acl' option in its configuration in
; a similar fashion to the usual 'permit' and 'deny' options. Example:
; acl=my_named_acl
;
; Multiple named ACLs can be applied by either comma separating the arguments or
; just by adding additional ACL lines. Example:
; acl=my_named_acl
; acl=my_named_acl2
;
; or
;
; acl=my_named_acl,my_named_acl2
;
; ACLs specified by name are evaluated independently from the ACL specified via
; permit/deny. In order for an address to pass a given ACL, it must pass both
; the ACL specified by permit/deny for a given item as well as any named ACLs
; that were specified.
;
;[example_named_acl1]
;deny=0.0.0.0/0.0.0.0
;permit=209.16.236.0
;permit=209.16.236.1
;
;[example_named_acl2]
;permit=0.0.0.0/0.0.0.0
;deny=10.24.20.171
;deny=10.24.20.103
;deny=209.16.236.1
;
; example_named_acl1 above shows an example of whitelisting. When whitelisting, the
; named ACLs should follow a deny that blocks everything (like deny=0.0.0.0/0.0.0.0)
; The following example explains how combining the ACLs works:
; <in another configuration>
; [example_item_with_acl]
; acl=example_named_acl1
; acl=example_named_acl2
;
; Suppose 209.16.236.0 tries to communicate and the ACL for that example is applied to it...
; First, example_named_acl1 is evaluated. The address is allowed by that ACL.
; Next, example_named_acl2 is evaluated. The address isn't blocked by example_named_acl2
; either, so it passes.
;
; Suppose instead 209.16.236.1 tries to communicate and the same ACL is applied.
; First, example_named_acl1 is evaluated and the address is allowed.
; However, it is blocked by example_named_acl2, so the address is blocked from the combined
; ACL.
;
; Similarly, the permits/denies in specific configurations that make up an ACL definition
; are also treated as a separate ACL for evaluation. So if we change the example above to:
; <in another configuration>
; [example_item_with_acl]
; acl=example_named_acl1
; acl=example_named_acl2
; deny=209.16.236.0
;
; Then 209.16.236.0 will be rejected by the non-named component of the combined ACL even
; though it passes the two named components.
;
;
; Named ACLs can use ipv6 addresses just like normal ACLs.
;[ipv6_example_1]
;deny = ::
;permit = ::1/128
;
;[ipv6_example_2]
;permit = fe80::21d:bad:fad:2323

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voice-client/files/etc/asterisk_templates/cdr_custom.conf.TEMPLATE Normal file
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;
; Mappings for custom config file
;
; To get your CSV output in a format tailored to your liking, uncomment the
; following lines and look for the output in the cdr-custom directory (usually
; in /var/log/asterisk). Depending on which mapping you uncomment, you may see
; Master.csv, Simple.csv, or both.
;
;[mappings]
;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)}
;
; High Resolution Time for billsec and duration fields
;Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration,f)})},${CSV_QUOTE(${CDR(billsec,f)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)}
;Simple.csv => ${CSV_QUOTE(${EPOCH})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})}

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voice-client/files/etc/asterisk_templates/cel.conf.TEMPLATE Normal file
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;
; Asterisk Channel Event Logging (CEL)
;
; Channel Event Logging is a mechanism to provide fine-grained event information
; that can be used to generate billing information. Such event information can
; be recorded to various backend modules.
;
[general]
; CEL Activation
;
; Use the 'enable' keyword to turn CEL on or off.
;
; Accepted values: yes and no
; Default value: no
;enable=yes
; Application Tracking
;
; Use the 'apps' keyword to specify the list of applications for which you want
; to receive CEL events. This is a comma separated list of Asterisk dialplan
; applications, such as Dial, Queue, and Park.
;
; Accepted values: A comma separated list of Asterisk dialplan applications
; Default value: none
;
; Note: You may also use 'all' which will result in CEL events being reported
; for all Asterisk applications. This may affect Asterisk's performance
; significantly.
apps=dial,park
; Event Tracking
;
; Use the 'events' keyword to specify the list of events which you want to be
; raised when they occur. This is a comma separated list of the values in the
; table below.
;
; Accepted values: A comma separated list of one or more of the following:
; ALL -- Generate entries on all events
; CHAN_START -- The time a channel was created
; CHAN_END -- The time a channel was terminated
; ANSWER -- The time a channel was answered (ie, phone taken off-hook)
; HANGUP -- The time at which a hangup occurred
; BRIDGE_ENTER -- The time a channel was connected into a conference room
; BRIDGE_EXIT -- The time a channel was removed from a conference room
; APP_START -- The time a tracked application was started
; APP_END -- the time a tracked application ended
; PARK_START -- The time a call was parked
; PARK_END -- Unpark event
; BLINDTRANSFER -- When a blind transfer is initiated
; ATTENDEDTRANSFER -- When an attended transfer is initiated
; PICKUP -- This channel picked up the specified channel
; FORWARD -- This channel is being forwarded somewhere else
; LINKEDID_END -- The last channel with the given linkedid is retired
; USER_DEFINED -- Triggered from the dialplan, and has a name given by the
; user
; LOCAL_OPTIMIZE -- A local channel pair is optimizing away.
;
; Default value: none
; (Track no events)
events=APP_START,CHAN_START,CHAN_END,ANSWER,HANGUP,BRIDGE_ENTER,BRIDGE_EXIT
; Date Format
;
; Use the 'dateformat' keyword to specify the date format used when CEL events
; are raised.
;
; Accepted values: A strftime format string (see man strftime)
;
; Example: "%F %T"
; -> This gives the date and time in the format "2009-06-23 17:02:35"
;
; If this option is not specified, the default format is "<seconds>.<microseconds>"
; since epoch. The microseconds field will always be 6 digits in length, meaning it
; may have leading zeros.
;
;dateformat = %F %T
;
; Asterisk Manager Interface (AMI) CEL Backend
;
[manager]
; AMI Backend Activation
;
; Use the 'enable' keyword to turn CEL logging to the Asterisk Manager Interface
; on or off.
;
; Accepted values: yes and no
; Default value: no
;enabled=yes
; Use 'show_user_defined' to put "USER_DEFINED" in the EventName header,
; instead of (by default) just putting the user defined event name there.
; When enabled the UserDefType header is added for user defined events to
; provide the user defined event name.
;
;show_user_defined=yes
;
; RADIUS CEL Backend
;
[radius]
;
; Log date/time in GMT
;usegmtime=yes
;
; Set this to the location of the radiusclient-ng configuration file
; The default is /etc/radiusclient-ng/radiusclient.conf
;radiuscfg => /usr/local/etc/radiusclient-ng/radiusclient.conf
;

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voice-client/files/etc/asterisk_templates/confbridge.conf.TEMPLATE Normal file
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[general]
; The general section of this config
; is not currently used, but reserved
; for future use.
;
; --- Default Information ---
; The default_user and default_bridge sections are applied
; automatically to all ConfBridge instances invoked without
; a user, or bridge argument. No menu is applied by default.
;
; Note that while properties of the default_user or default_bridge
; profile can be overridden, if removed, they will be automatically
; added and made available to the dialplan upon module load.
;
; --- ConfBridge User Profile Options ---
[default_user]
type=user
;admin=yes ; Sets if the user is an admin or not. Off by default.
;marked=yes ; Sets if this is a marked user or not. Off by default.
;startmuted=yes; Sets if all users should start out muted. Off by default
;music_on_hold_when_empty=yes ; Sets whether MOH should be played when only
; one person is in the conference or when the
; the user is waiting on a marked user to enter
; the conference. Off by default.
;music_on_hold_class=default ; The MOH class to use for this user.
;quiet=yes ; When enabled enter/leave prompts and user intros are not played.
; There are some prompts, such as the prompt to enter a PIN number,
; that must be played regardless of what this option is set to.
; Off by default
;announce_user_count=yes ; Sets if the number of users should be announced to the
; caller. Off by default.
;announce_user_count_all=yes ; Sets if the number of users should be announced to
; all the other users in the conference when someone 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.
;announce_only_user=yes ; Sets if the only user announcement should be played
; when a channel enters a empty conference. On by default.
;wait_marked=yes ; Sets if the user must wait for a marked user to enter before
; joining the conference. Off by default.
;end_marked=yes ; This option will kick every user with this option set in their
; user profile after the last Marked user exists the conference.
;dsp_drop_silence=yes ; 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.
;dsp_talking_threshold=128 ; 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.
;
; What this value affects internally:
;
; 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.
; 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.
; 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.
;
; By default this value is 160 ms. Valid values are 1 through 2^31
;dsp_silence_threshold=2000 ; 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.
;
; What this value affects internally:
;
; 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.
; 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.
;
; 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.
;
; By default this value is 2500ms. Valid values are 1 through 2^31
;talk_detection_events=yes ; This option sets whether or not notifications of when a user
; begins and ends talking should be sent out as events over AMI.
; By default this option is off.
;denoise=yes ; Sets whether or not a denoise filter should be applied
; to the audio before mixing or not. Off by default. Requires
; func_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.
;jitterbuffer=yes ; 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 JITTERBUFFER
; dialplan function's default adaptive jitterbuffer. For a more fine tuned
; jitterbuffer, disable this option and use the JITTERBUFFER dialplan function
; on the user before entering the ConfBridge application.
;pin=1234 ; Sets if this user must enter a PIN number before entering
; the conference. The PIN will be prompted for.
;announce_join_leave=yes ; When enabled, this option will prompt the user for a
; name when entering the conference. After the name is
; recorded, it will be played as the user enters and exists
; the conference. This option is off by default.
;announce_join_leave_review=yes ; When enabled, implies announce_join_leave, but the user
; will be prompted to review their recording before
; entering the conference. During this phase, the recording
; may be listened to, re-recorded, or accepted as is. This
; option is off by default.
;dtmf_passthrough=yes ; Sets whether or not DTMF should pass through the conference.
; This option is off by default.
;announcement=</path/to/file> ; Play a sound file to the user when they join the conference.
;timeout=3600 ; When set non-zero, this specifies the number of seconds that the participant
; may stay in the conference before being automatically ejected. When the user
; is ejected from the conference, the user's channel will have the CONFBRIDGE_RESULT
; variable set to "TIMEOUT". A value of 0 indicates that there is no timeout.
; Default: 0
; --- ConfBridge Bridge Profile Options ---
[default_bridge]
type=bridge
;max_members=50 ; 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.
;record_conference=yes ; 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
; 'confbridge-<name of conference bridge>-<start time>.wav
; and the default format is 8khz slinear. This file will be
; located in the configured monitoring directory in asterisk.conf.
;record_file=</path/to/file> ; 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 CONFBRIDGE() 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.
;internal_sample_rate=auto ; 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.
;mixing_interval=40 ; 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 40ms 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. By default 20ms is used.
;video_mode = follow_talker; 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.
;
; --- MODES ---
; none: 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.
;
; follow_talker: The video feed will follow whoever is talking and providing video.
;
; last_marked: 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.
;
; first_marked: 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.
;language=en ; Set the language used for announcements to the conference.
; Default is en (English).
;regcontext=conferences ; The name of the context into which to register conference names as extensions.
; 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: sound_had_joined=conf-hasjoin This will play the conf-hasjoin
; sound file found in the sounds directory when announcing someone's name is joining the
; conference.
;sound_join ; The sound played to everyone when someone enters the conference.
;sound_leave ; The sound played to everyone when someone leaves the conference.
;sound_has_joined ; The sound played before announcing someone's name has
; joined the conference. This is used for user intros.
; Example "_____ has joined the conference"
;sound_has_left ; The sound played when announcing someone's name has
; left the conference. This is used for user intros.
; Example "_____ has left the conference"
;sound_kicked ; The sound played to a user who has been kicked from the conference.
;sound_muted ; The sound played when the mute option it toggled on.
;sound_unmuted ; The sound played when the mute option it toggled off.
;sound_only_person ; The sound played when the user is the only person in the conference.
;sound_only_one ; The sound played to a user when there is only one other
; person is in the conference.
;sound_there_are ; The sound played when announcing how many users there
; are in a conference.
;sound_other_in_party; ; This file is used in conjunction with 'sound_there_are"
; when announcing how many users there are in the conference.
; The sounds are stringed together like this.
; "sound_there_are" <number of participants> "sound_other_in_party"
;sound_place_into_conference ; The sound played when someone is placed into the conference
; after waiting for a marked user. This sound is now deprecated
; since it was only ever used improperly and correcting that bug
; made it completely unused.
;sound_wait_for_leader ; The sound played when a user is placed into a conference that
; can not start until a marked user enters.
;sound_leader_has_left ; The sound played when the last marked user leaves the conference.
;sound_get_pin ; The sound played when prompting for a conference pin number.
;sound_invalid_pin ; The sound played when an invalid pin is entered too many times.
;sound_locked ; The sound played to a user trying to join a locked conference.
;sound_locked_now ; The sound played to an admin after toggling the conference to locked mode.
;sound_unlocked_now; The sound played to an admin after toggling the conference to unlocked mode.
;sound_error_menu ; The sound played when an invalid menu option is entered.
;sound_begin ; The sound played to the conference when the first marked user enters the conference.
; --- ConfBridge Menu Options ---
; 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.
;
; Below is a list of menu actions that can be assigned
; to a DTMF sequence.
;
; To have the first DTMF digit in a sequence be the '#' character, you need to
; escape it. If it is not escaped then normal config file processing will
; think it is a directive like #include. For example:
; \#1=toggle_mute ; Pressing #1 will toggle the mute setting.
;
; A single DTMF sequence can have multiple actions associated with it. This is
; accomplished by stringing the actions together and using a ',' as the delimiter.
; Example: Both listening and talking volume is reset when '5' is pressed.
; 5=reset_talking_volume, reset_listening_volume
;
; playback(<name of audio file>&<name of audio file>)
; Playback will play back an audio file to a channel
; and then immediately return to the conference.
; This file can not be interupted by DTMF.
; Mutliple files can be chained together using the
; '&' character.
; playback_and_continue(<name of playback prompt>&<name of playback prompt>)
; playback_and_continue 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 '&' character as a delimiter.
; toggle_mute ; 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.
; no_op ; 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.
; decrease_listening_volume ; Decreases the channel's listening volume.
; increase_listening_volume ; Increases the channel's listening volume.
; reset_listening_volume ; Reset channel's listening volume to default level.
; decrease_talking_volume ; Decreases the channel's talking volume.
; increase_talking_volume ; Icreases the channel's talking volume.
; reset_talking_volume ; Reset channel's talking volume to default level.
;
; dialplan_exec(context,exten,priority) ; The dialplan_exec 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!
; leave_conference ; This action allows a user to exit the conference and continue
; execution in the dialplan.
;
; admin_kick_last ; 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.
;
; admin_toggle_conference_lock ; 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.
; set_as_single_video_src ; 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 video_mode is set to.
; release_as_single_video_src ; This action allows a user to release themselves as
; the video source. If video_mode is not set to "none"
; this action will result in the conference returning to
; whatever video mode the bridge profile is using.
;
; 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 video_mode option is set to
; upon release of the video src.
; admin_toggle_mute_participants ; This action allows an administrator to toggle the mute
; state for all non-admins within a conference.
; Subsequent non-admins joining a muted conference will
; start muted. All admin users are unaffected by this
; option. Note that all users, regardless of their admin
; status, are notified that the conference is muted when
; the state is toggled.
; participant_count ; This action plays back the number of participants currently
; in a conference
[sample_user_menu]
type=menu
*=playback_and_continue(conf-usermenu)
*1=toggle_mute
1=toggle_mute
*4=decrease_listening_volume
4=decrease_listening_volume
*6=increase_listening_volume
6=increase_listening_volume
*7=decrease_talking_volume
7=decrease_talking_volume
*8=leave_conference
8=leave_conference
*9=increase_talking_volume
9=increase_talking_volume
[sample_admin_menu]
type=menu
*=playback_and_continue(conf-adminmenu)
*1=toggle_mute
1=toggle_mute
*2=admin_toggle_conference_lock ; only applied to admin users
2=admin_toggle_conference_lock ; only applied to admin users
*3=admin_kick_last ; only applied to admin users
3=admin_kick_last ; only applied to admin users
*4=decrease_listening_volume
4=decrease_listening_volume
*6=increase_listening_volume
6=increase_listening_volume
*7=decrease_talking_volume
7=decrease_talking_volume
*8=no_op
8=no_op
*9=increase_talking_volume
9=increase_talking_volume

26
voice-client/files/etc/asterisk_templates/udptl.conf.TEMPLATE Normal file
View file

@ -0,0 +1,26 @@
;
; UDPTL Configuration (UDPTL is one of the transports for T.38)
;
[general]
;
; UDPTL start and UDPTL end configure start and end addresses
;
udptlstart=4000
udptlend=4999
;
; Whether to enable or disable UDP checksums on UDPTL traffic
;
;udptlchecksums=no
;
; The number of error correction entries in a UDPTL packet
;
udptlfecentries = 3
;
; The span over which parity is calculated for FEC in a UDPTL packet
;
udptlfecspan = 3
;
; Some VoIP providers will only accept an offer with an even-numbered
; UDPTL port. Set this option so that Asterisk will only attempt to use
; even-numbered ports when negotiating T.38. Default is no.
use_even_ports = no

9
voice-client/files/etc/init.d/voice_client
View file

@ -67,6 +67,7 @@ TMPL_QUEUE=$TEMPLATEDIR/queue.TEMPLATE
TMPL_ASTERISK=$TEMPLATEDIR/asterisk.conf.TEMPLATE
TMPL_CDR=$TEMPLATEDIR/cdr.conf.TEMPLATE
TMPL_CDR_SYSLOG=$TEMPLATEDIR/cdr_syslog.conf.TEMPLATE
TMPL_CDR_CUSTOM=$TEMPLATEDIR/cdr_custom.conf.TEMPLATE
TMPL_INDICATIONS=$TEMPLATEDIR/indications.conf.TEMPLATE
TMPL_CEL=$TEMPLATEDIR/cel.conf.TEMPLATE
TMPL_LOGGER=$TEMPLATEDIR/logger.conf.TEMPLATE
@ -75,6 +76,9 @@ TMPL_MODULES=$TEMPLATEDIR/modules.conf.TEMPLATE
TMPL_FEATURES=$TEMPLATEDIR/features.conf.TEMPLATE
TMPL_CODECS=$TEMPLATEDIR/codecs.conf.TEMPLATE
TMPL_MUSICONHOLD=$TEMPLATEDIR/musiconhold.conf.TEMPLATE
TMPL_ACL=$TEMPLATEDIR/acl.conf.TEMPLATE
TMPL_CONFBRIDGE=$TEMPLATEDIR/confbridge.conf.TEMPLATE
TMPL_UDPTL=$TEMPLATEDIR/udptl.conf.TEMPLATE
TMPL_BRCM=$TEMPLATEDIR/brcm.conf.TEMPLATE
TMPL_BRCM_LINE=$TEMPLATEDIR/brcm_line.TEMPLATE
@ -150,6 +154,11 @@ assemble_and_copy_config()
[ -f $TMPL_MODULES ] && cp $TMPL_MODULES $WORKDIR/modules.conf
[ -f $TMPL_EXTENSIONS_MACRO ] && cp $TMPL_EXTENSIONS_MACRO $WORKDIR/extensions_macro.conf
[ -f $TMPL_MUSICONHOLD ] && cp $TMPL_MUSICONHOLD $WORKDIR/musiconhold.conf
[ -f $TMPL_ACL ] && cp $TMPL_ACL $WORKDIR/acl.conf
[ -f $TMPL_CDR_CUSTOM ] && cp $TMPL_CDR_CUSTOM $WORKDIR/cdr_custom.conf
[ -f $TMPL_CDR_SYSLOG ] && cp $TMPL_CDR_SYSLOG $WORKDIR/cdr_syslog.conf
[ -f $TMPL_CONFBRIDGE ] && cp $TMPL_CONFBRIDGE $WORKDIR/confbridge.conf
[ -f $TMPL_UDPTL ] && cp $TMPL_UDPTL $WORKDIR/udptl.conf
[ -f $SPECRATECFG ] && cp $SPECRATECFG $WORKDIR/special_rate_nr.cfg
test -e $TMPL_MEETME && cp $TMPL_MEETME $WORKDIR/meetme.conf