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authorPatrick McHardy <kaber@trash.net>2011-06-08 14:20:40 +0200
committerPatrick McHardy <kaber@trash.net>2011-06-08 14:20:40 +0200
commit84c94e92c153057470194aadf7ae22a2569f1bd4 (patch)
tree122c43bc5c85c2e2a81b2ab6cd7d59d9d71c4475 /configs
parent2ac6b3ac85972f5d4ec2b82eb299a7d85ff91c92 (diff)
parent70f4b1801b76660aed3487fa304e71f872daa077 (diff)
Merge 192.168.0.100:/repos/git/asterisk
Diffstat (limited to 'configs')
-rw-r--r--configs/ccss.conf.sample44
-rw-r--r--configs/cel.conf.sample2
-rw-r--r--configs/cel_pgsql.conf.sample2
-rw-r--r--configs/chan_dahdi.conf.sample38
-rw-r--r--configs/confbridge.conf.sample302
-rw-r--r--configs/extensions.lua.sample14
-rw-r--r--configs/features.conf.sample3
-rw-r--r--configs/http.conf.sample5
-rw-r--r--configs/iax.conf.sample2
-rw-r--r--configs/jabber.conf.sample3
-rw-r--r--configs/queues.conf.sample2
-rw-r--r--configs/sip.conf.sample30
-rw-r--r--configs/sip_notify.conf.sample2
-rw-r--r--configs/skinny.conf.sample9
-rw-r--r--configs/users.conf.sample2
15 files changed, 449 insertions, 11 deletions
diff --git a/configs/ccss.conf.sample b/configs/ccss.conf.sample
index bb78cad..2636f7e 100644
--- a/configs/ccss.conf.sample
+++ b/configs/ccss.conf.sample
@@ -6,12 +6,54 @@
;
[general]
-; There is only a single option that may be defined in this file.
; The cc_max_requests option is a global limit on the number of
; CC requests that may be in the Asterisk system at any time.
;
;cc_max_requests = 20
;
+; The cc_STATE_devstate variables listed below can be used to change the
+; default mapping of the internal state machine tracking the state of
+; call completion to an Asterisk Device State value. The acceptable values
+; that can be provided are as follows, with a description of what the
+; equivalent device BLF that this maps to:
+;
+; UNKNOWN ; Device is valid but channel didn't know state
+; NOT_INUSE ; Device is not used
+; INUSE ; Device is in use
+; BUSY ; Device is busy
+; INVALID ; Device is invalid
+; UNAVAILABLE ; Device is unavailable
+; RINGING ; Device is ringing
+; RINGINUSE ; Device is ringing *and* in use
+; ONHOLD ; Device is on hold
+;
+; These states are used to generate DEVICE_STATE information that can be
+; included with Asterisk hints for phones to subscribe to the state information
+; or dialplan to check the state using the EXTENSION_STATE() function or
+; the DEVICE_STATE() function.
+;
+; The states are in the format of: "ccss:TECH/ID" so an example of device
+; SIP/3000 making a CallCompletionRequest() could be checked by looking at
+; DEVICE_STATE(ccss:SIP/3000) or an Asterisk Hint could be generated such as
+;
+; [hint-context]
+; exten => *843000,hint,ccss:SIP/3000
+;
+; and then accessed with EXTENSION_STATE(*843000@hint-context)
+; or subscribed to with a BLF button on a phone.
+;
+; The available state mapping and default values are:
+;
+; cc_available_devstate = NOT_INUSE
+; cc_offered_devstate = NOT_INUSE
+; cc_caller_requested_devstate = NOT_INUSE
+; cc_active_devstate = INUSE
+; cc_callee_ready_devstate = INUSE
+; cc_caller_busy_devstate = ONHOLD
+; cc_recalling_devstate = RINGING
+; cc_complete_devstate = NOT_INUSE
+; cc_failed_devstate = NOT_INUSE
+
;
;============================================
; PLEASE READ THIS!!!
diff --git a/configs/cel.conf.sample b/configs/cel.conf.sample
index 65d79cd..d9ba90c 100644
--- a/configs/cel.conf.sample
+++ b/configs/cel.conf.sample
@@ -4,7 +4,7 @@
; 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 databases and files via pluggable backend modules.
+; be recorded to various backend modules.
;
[general]
diff --git a/configs/cel_pgsql.conf.sample b/configs/cel_pgsql.conf.sample
index 7588211..67d0574 100644
--- a/configs/cel_pgsql.conf.sample
+++ b/configs/cel_pgsql.conf.sample
@@ -51,7 +51,7 @@
; amaflag (an int)
; userfield
; peer
-
+; extra
[global]
;hostname=localhost
diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample
index 761c576..16269dc 100644
--- a/configs/chan_dahdi.conf.sample
+++ b/configs/chan_dahdi.conf.sample
@@ -236,6 +236,19 @@
;
;mcid_send=yes
+; Send ISDN date/time IE in CONNECT message option. Only valid on NT spans.
+;
+; no: Do not send date/time IE in CONNECT message.
+; date: Send date only.
+; date_hh Send date and hour.
+; date_hhmm Send date, hour, and minute.
+; date_hhmmss Send date, hour, minute, and second.
+;
+; Default is an empty string which lets libpri pick the default
+; date/time IE send policy.
+;
+;datetime_send=
+
; Allow inband audio (progress) when a call is DISCONNECTed by the far end of a PRI
;
;inbanddisconnect=yes
@@ -581,13 +594,28 @@ callwaiting=yes
; Allow incoming ISDN call waiting calls.
; A call waiting call is a SETUP message with no B channel selected.
;allow_call_waiting_calls=no
-;
+
; Configure the ISDN span to indicate MWI for the list of mailboxes.
; You can give a comma separated list of up to 8 mailboxes per span.
; An empty list disables MWI.
; The default is an empty list.
;mwi_mailboxes=mailbox_number[@context]{,mailbox_number[@context]}
;
+; Configure the ISDN span voicemail numbers for MWI mailboxes. What number
+; to call for a user to retrieve voicemail messages.
+;
+; You can give a comma separated list of numbers. The position of the number
+; corresponds to the position in mwi_mailboxes. If a position is empty then
+; the last number is reused.
+;
+; For example:
+; mwi_vm_numbers=700,,800,,900
+; is equivalent to:
+; mwi_vm_numbers=700,700,800,800,900
+;
+; The default is no number.
+;mwi_vm_numbers=
+
; Whether or not restrict outgoing caller ID (will be sent as ANI only, not
; available for the user)
; Mostly use with FXS ports
@@ -820,6 +848,11 @@ pickupgroup=1
;
;useincomingcalleridondahditransfer = yes
;
+; Add a description for the channel which can be shown through the Asterisk
+; console when executing the 'dahdi show channels' command is run.
+;
+;description=Phone located in lobby
+;
; AMA flags affects the recording of Call Detail Records. If specified
; it may be 'default', 'omit', 'billing', or 'documentation'.
;
@@ -1068,10 +1101,13 @@ pickupgroup=1
;
;
;callerid="Green Phone"<(256) 428-6121>
+;description=Reception Phone ; add a description for 'dahdi show channels'
;channel => 1
;callerid="Black Phone"<(256) 428-6122>
+;description=Courtesy Phone
;channel => 2
;callerid="CallerID Phone" <(630) 372-1564>
+;description= ; reset the description for following channels
;channel => 3
;callerid="Pac Tel Phone" <(256) 428-6124>
;channel => 4
diff --git a/configs/confbridge.conf.sample b/configs/confbridge.conf.sample
new file mode 100644
index 0000000..1781b88
--- /dev/null
+++ b/configs/confbridge.conf.sample
@@ -0,0 +1,302 @@
+[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.
+;
+
+; --- 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
+ ; 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.
+
+;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.
+;dtmf_passthrough=yes ; Sets whether or not DTMF should pass through the conference.
+ ; This option is off by default.
+
+; --- 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.
+
+; 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.
+;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.
+
+; --- 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.
+;
+; 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.
+ ; continue to collect the dtmf sequence.
+; 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.
+
+[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=no_op
+8=no_op
+*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
diff --git a/configs/extensions.lua.sample b/configs/extensions.lua.sample
index bd6fb1f..5868de4 100644
--- a/configs/extensions.lua.sample
+++ b/configs/extensions.lua.sample
@@ -88,12 +88,14 @@ TRUNKMSD = 1
--
-- More examples can be found below.
--
--- Before starting long running operations, an autoservice should be started
--- using the autoservice_start() function. This autoservice will automatically
--- be stopped before executing applications and dialplan functions and will be
--- restarted afterwards. The autoservice can be stopped using
--- autoservice_stop() and the autoservice_status() function will return true if
--- an autoservice is currently running.
+-- An autoservice is automatically run while lua code is executing. The
+-- autoservice can be stopped and restarted using the autoservice_stop() and
+-- autoservice_start() functions. The autservice should be running before
+-- starting long running operations. The autoservice will automatically be
+-- stopped before executing applications and dialplan functions and will be
+-- restarted afterwards. The autoservice_status() function can be used to
+-- check the current status of the autoservice and will return true if an
+-- autoservice is currently running.
--
function outgoing_local(c, e)
diff --git a/configs/features.conf.sample b/configs/features.conf.sample
index 7534d16..f9d9dd4 100644
--- a/configs/features.conf.sample
+++ b/configs/features.conf.sample
@@ -83,6 +83,9 @@ context => parkedcalls ; Which context parked calls are in (default parking lot
; You can set parkinglot with the CHANNEL dialplan function
; or by setting 'parkinglot' directly in the channel configuration file.
;
+; (Note: Leading '0's and any non-numerical characters on parkpos extensions
+; will be ignored. Parkext on the other hand can be any string.)
+;
;[parkinglot_edvina]
;context => edvinapark
;parkext => 799
diff --git a/configs/http.conf.sample b/configs/http.conf.sample
index f328ea6..8a63148 100644
--- a/configs/http.conf.sample
+++ b/configs/http.conf.sample
@@ -34,6 +34,11 @@ bindaddr=127.0.0.1
;
;prefix=asterisk
;
+; sessionlimit specifies the maximum number of httpsessions that will be
+; allowed to exist at any given time. (default: 100)
+;
+;sessionlimit=100
+;
; Whether Asterisk should serve static content from http-static
; Default is no.
;
diff --git a/configs/iax.conf.sample b/configs/iax.conf.sample
index a3d0fea..d08aa31 100644
--- a/configs/iax.conf.sample
+++ b/configs/iax.conf.sample
@@ -504,6 +504,7 @@ type=peer
username=asterisk
secret=supersecret
host=216.207.245.47
+description=Demo System At Digium ; Description of this peer, as listed by 'iax2 show peers'
;sendani=no
;host=asterisk.linux-support.net
;port=5036
@@ -544,6 +545,7 @@ host=216.207.245.47
;[biggateway]
;type=peer
;host=192.168.0.1
+;description=Gateway to PSTN
;context=*
;secret=myscret
;trunk=yes ; Use IAX2 trunking with this host
diff --git a/configs/jabber.conf.sample b/configs/jabber.conf.sample
index 098122d..a838568 100644
--- a/configs/jabber.conf.sample
+++ b/configs/jabber.conf.sample
@@ -34,3 +34,6 @@
; Messages stored longer than this value will be deleted by Asterisk.
; This option applies to incoming messages only, which are intended to
; be processed by the JABBER_RECEIVE dialplan function.
+;sendtodialplan=yes ; Send incoming messages into the dialplan. Off by default.
+;context=messages ; Dialplan context to send incoming messages to. If not set,
+ ; "default" will be used.
diff --git a/configs/queues.conf.sample b/configs/queues.conf.sample
index cb70dde..c2045f9 100644
--- a/configs/queues.conf.sample
+++ b/configs/queues.conf.sample
@@ -318,6 +318,8 @@ monitor-type = MixMonitor
;queue-callswaiting = queue-callswaiting
; ("The current est. holdtime is")
;queue-holdtime = queue-holdtime
+ ; ("minute.")
+;queue-minute = queue-minute
; ("minutes.")
;queue-minutes = queue-minutes
; ("seconds.")
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index a60ea34..49277d6 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -202,6 +202,16 @@ tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0
; For details how to construct a certificate for SIP see
; http://tools.ietf.org/html/draft-ietf-sip-domain-certs
+;tcpauthtimeout = 30 ; tcpauthtimeout specifies the maximum number
+ ; of seconds a client has to authenticate. If
+ ; the client does not authenticate beofre this
+ ; timeout expires, the client will be
+ ; disconnected. (default: 30 seconds)
+
+;tcpauthlimit = 100 ; tcpauthlimit specifies the maximum number of
+ ; unauthenticated sessions that will be allowed
+ ; to connect at any given time. (default: 100)
+
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
@@ -375,6 +385,16 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;auth_options_requests = yes ; Enabling this option will authenticate OPTIONS requests just like
; INVITE requests are. By default this option is disabled.
+;accept_outofcall_message = no ; Disable this option to reject all MESSAGE requests outside of a
+ ; call. By default, this option is enabled. When enabled, MESSAGE
+ ; requests are passed in to the dialplan.
+
+;auth_message_requests = yes ; Enabling this option will authenticate MESSAGE requests.
+ ; By default this option is enabled. However, it can be disabled
+ ; should an application desire to not load the Asterisk server with
+ ; doing authentication and implement end to end security in the
+ ; message body.
+
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
; order instead of RFC3551 packing order (this is required
; for Sipura and Grandstream ATAs, among others). This is
@@ -422,6 +442,14 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; If you have qualify on and the peer becomes unreachable
; this setting will enforce inactivation of the regexten
; extension for the peer
+;legacy_useroption_parsing=yes ; Default "no" ; If you have this option enabled and there are semicolons
+ ; in the user field of a sip URI, the field be truncated
+ ; at the first semicolon seen. This effectively makes
+ ; semicolon a non-usable character for peer names, extensions,
+ ; and maybe other, less tested things. This can be useful
+ ; for improving compatability with devices that like to use
+ ; user options for whatever reason. The behavior is similar to
+ ; how SIP URI's were typically handled in 1.6.2, hence the name.
; The shrinkcallerid function removes '(', ' ', ')', non-trailing '.', and '-' not
; in square brackets. For example, the caller id value 555.5555 becomes 5555555
@@ -1100,6 +1128,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; use_q850_reason
; maxforwards
; encryption
+; description ; Used to provide a description of the peer in console output
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
@@ -1195,6 +1224,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;context=from-sip ; Where to start in the dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override the phones config
; on incoming calls to Asterisk
+;description=Courtesy Phone ; Description of the peer. Shown when doing 'sip show peers'.
;host=192.168.0.23 ; we have a static but private IP address
; No registration allowed
;nat=no ; there is not NAT between phone and Asterisk
diff --git a/configs/sip_notify.conf.sample b/configs/sip_notify.conf.sample
index 3b1a65d..8d49126 100644
--- a/configs/sip_notify.conf.sample
+++ b/configs/sip_notify.conf.sample
@@ -44,7 +44,7 @@ Event=>report
Event=>check-sync\;reboot=false
[snom-reboot]
-Event=>reboot
+Event=>check-sync\;reboot=true
; Cisco
diff --git a/configs/skinny.conf.sample b/configs/skinny.conf.sample
index 2199af1..d40823e 100644
--- a/configs/skinny.conf.sample
+++ b/configs/skinny.conf.sample
@@ -9,6 +9,15 @@ dateformat=M-D-Y ; M,D,Y in any order (6 chars max)
; Use M for month, D for day, Y for year, A for 12-hour time.
keepalive=120
+;authtimeout = 30 ; authtimeout specifies the maximum number of seconds a
+ ; client has to authenticate. If the client does not
+ ; authenticate beofre this timeout expires, the client
+ ; will be disconnected. (default: 30 seconds)
+
+;authlimit = 50 ; authlimit specifies the maximum number of
+ ; unauthenticated sessions that will be allowed to
+ ; connect at any given time. (default: 50)
+
;vmexten=8500 ; Systemwide voicemailmain pilot number
; It must be in the same context as the calling
; device/line
diff --git a/configs/users.conf.sample b/configs/users.conf.sample
index 7612546..50b80a1 100644
--- a/configs/users.conf.sample
+++ b/configs/users.conf.sample
@@ -87,6 +87,8 @@ pickupgroup = 1
;[6000]
;fullname = Joe User
+;description = Courtesy Phone In Lobby ; Used to provide a description of the
+ ; peer in console output
;email = joe@foo.bar
;secret = 1234
;dahdichan = 1