diff options
Diffstat (limited to 'configs')
| -rw-r--r-- | configs/ccss.conf.sample | 44 | ||||
| -rw-r--r-- | configs/cel.conf.sample | 2 | ||||
| -rw-r--r-- | configs/cel_pgsql.conf.sample | 2 | ||||
| -rw-r--r-- | configs/chan_dahdi.conf.sample | 38 | ||||
| -rw-r--r-- | configs/confbridge.conf.sample | 302 | ||||
| -rw-r--r-- | configs/extensions.lua.sample | 14 | ||||
| -rw-r--r-- | configs/features.conf.sample | 3 | ||||
| -rw-r--r-- | configs/http.conf.sample | 5 | ||||
| -rw-r--r-- | configs/iax.conf.sample | 2 | ||||
| -rw-r--r-- | configs/jabber.conf.sample | 3 | ||||
| -rw-r--r-- | configs/queues.conf.sample | 2 | ||||
| -rw-r--r-- | configs/sip.conf.sample | 30 | ||||
| -rw-r--r-- | configs/sip_notify.conf.sample | 2 | ||||
| -rw-r--r-- | configs/skinny.conf.sample | 9 | ||||
| -rw-r--r-- | configs/users.conf.sample | 2 |
15 files changed, 449 insertions, 11 deletions
diff --git a/configs/ccss.conf.sample b/configs/ccss.conf.sample index bb78cad0c..2636f7ec9 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 65d79cdff..d9ba90cb5 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 75882118a..67d0574ab 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 761c5760b..16269dc39 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 000000000..1781b88a0 --- /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 bd6fb1f95..5868de4f7 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 7534d1616..f9d9dd45d 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 f328ea619..8a63148ff 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 a3d0fea12..d08aa31dc 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 098122d91..a83856867 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 cb70dde3f..c2045f90c 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 a60ea347d..49277d64f 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 3b1a65d07..8d4912647 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 2199af19d..d40823ef7 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 7612546b3..50b80a1c5 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 |