From d46f90058041a56d6ff4847ac2749b05a674fff5 Mon Sep 17 00:00:00 2001 From: lmadsen Date: Wed, 13 Apr 2011 15:49:33 +0000 Subject: Add 'description' field for CLI and Manager output (closes issue #19076) Reported by: lmadsen Patches: __20110408-channel-description.txt uploaded by lmadsen (license 10) Tested by: lmadsen Review: https://reviewboard.asterisk.org/r/1163/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313528 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/chan_dahdi.conf.sample | 8 ++++++++ configs/iax.conf.sample | 2 ++ configs/sip.conf.sample | 2 ++ configs/users.conf.sample | 2 ++ 4 files changed, 14 insertions(+) (limited to 'configs') diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample index 761c5760b..dca09cb56 100644 --- a/configs/chan_dahdi.conf.sample +++ b/configs/chan_dahdi.conf.sample @@ -820,6 +820,11 @@ 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 +1073,13 @@ 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/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/sip.conf.sample b/configs/sip.conf.sample index a60ea347d..5c8fa4de9 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -1100,6 +1100,7 @@ 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 +1196,7 @@ 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/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 -- cgit v1.2.3 From 6f2f7af1001dd2e6dc2ee0ff2498b32e9975255d Mon Sep 17 00:00:00 2001 From: rmudgett Date: Thu, 14 Apr 2011 18:22:35 +0000 Subject: Add Device State Information CCSS for Generic Devices. Add Asterisk Device State information and callbacks to the Call Completion Supplemental Services for generic agents. There are currently not many devices that have native support for CCSS. Even as the devices become available there may be other reasons why one may choose to not take advantage of the native abilities and stick with the generic implementation. The generic implementation is quite capable and could be greatly enhanced by adding device state capabilities. A phone could then subscribe to the device state with a BLF key in conjunction with Asterisk hints. The advantages of the device state information would allow a single button to: request CCSS, cancel a CCSS request, and display the current state of a CCSS request. For example, you may have a single button that when not lit, there is no active CCSS request. When you press that button, the dialplan can query the DEVICE_STATE() associated with that caller to determine whether they should be calling CallCompletionRequest() or CallCompletionCancel(). If there is currently a pending request, then the dialplan would cancel it. This also has the advantage of showing the true state of a request, which is an asynchronous call, even when CallCompletionRequest() thinks it was successful. The actual request could ultimately fail. Once lit, further feedback can be provided to the caller about the current state of their request since it will be updated by the CCSS State Machine as appropriate. The DEVICE_STATE mapping is configurable since the BLF being used on a given phone type may vary. The idea is to allow some level of customization as to the phone's behavior. As an example, you may want the BLF key to go solid once you have requested a callback. You may then want the LED to blink (typically ringing) when either the callback is in process, which is a visual indication that the incoming call is the desired callback. You may want it to blink when the callee is ready but you are busy, giving you a visual indication that the target is available as you may want to get off the line so that the callback can be successful. Device state information is sent back via the ast_devstate_prov_add() callback for any generic CCSS device as it traverses through the state machine. You simply provide a map between CC_STATE values and the corresponding AST_DEVICE state values. You could then generate hints against these states similar to what is possible today with Custom Devstates or MeetMe states. For example, you may have an extension 3000 that is currently associated with device SIP/3000. You could then create a feature code for that extension that may look something like: exten => *823000,hint,ccss:sip/3000 You would then subscribe a BLF button to *823000 which would point to the dialplan that handled CCSS requests/cancels using the available DEVICE_STATE() information about ccss:sip/3000 to make the decision about what to do. (closes issue #18788) Reported by: p_lindheimer Patches: ccss.trunk.18788.patch uploaded by p lindheimer (license 558) Modified with final reviewboard comments. Tested by: p_lindheimer, loloski Review: https://reviewboard.asterisk.org/r/1105/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@313744 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/ccss.conf.sample | 44 +++++++++++++++++++++++++++++++++++++++++++- 1 file changed, 43 insertions(+), 1 deletion(-) (limited to 'configs') 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!!! -- cgit v1.2.3 From 55d93db9b222ce832556e90ab36113b99b689a58 Mon Sep 17 00:00:00 2001 From: rmudgett Date: Mon, 18 Apr 2011 19:48:00 +0000 Subject: Problems with ISDN MWI to phones. The "controlling user number" is always the number of the voice mail box which is identical with the subscriber number itself. This number which is listed in the ISDN phone MWI menu cannot be called back to contact the voice mail box. The controlling user number should be made configurable. JIRA ABE-2738 JIRA SWP-2846 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314116 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/chan_dahdi.conf.sample | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'configs') diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample index dca09cb56..604f91708 100644 --- a/configs/chan_dahdi.conf.sample +++ b/configs/chan_dahdi.conf.sample @@ -581,13 +581,28 @@ 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 -- cgit v1.2.3 From c7b7b920afe4afccb39c7ea3c8d0bd4e56ad41f9 Mon Sep 17 00:00:00 2001 From: dvossel Date: Thu, 21 Apr 2011 18:11:40 +0000 Subject: New HD ConfBridge conferencing application. Includes a new highly optimized and customizable ConfBridge application capable of mixing audio at sample rates ranging from 8khz-192khz. Review: https://reviewboard.asterisk.org/r/1147/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314598 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/confbridge.conf.sample | 302 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 302 insertions(+) create mode 100644 configs/confbridge.conf.sample (limited to 'configs') 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--.wav + ; and the default format is 8khz slinear. This file will be + ; located in the configured monitoring directory in asterisk.conf. + +;record_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" "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(&) + ; 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(&) + ; 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 -- cgit v1.2.3 From 1c24e78eae7a8f5a45081c141e5f7387a623017e Mon Sep 17 00:00:00 2001 From: mnicholson Date: Thu, 21 Apr 2011 18:32:50 +0000 Subject: Merged revisions 314628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314628 | mnicholson | 2011-04-21 13:24:05 -0500 (Thu, 21 Apr 2011) | 27 lines Merged revisions 314620 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314620 | mnicholson | 2011-04-21 13:22:19 -0500 (Thu, 21 Apr 2011) | 20 lines Merged revisions 314607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r314607 | mnicholson | 2011-04-21 13:19:21 -0500 (Thu, 21 Apr 2011) | 14 lines Added limits to the number of unauthenticated sessions TCP based protocols are allowed to have open simultaneously. Also added timeouts for unauthenticated sessions where it made sense to do so. Unrelated, the manager interface now properly checks if the user has the "system" privilege before executing shell commands via the Originate action. AST-2011-005 AST-2011-006 (closes issue #18787) Reported by: kobaz (related to issue #18996) Reported by: tzafrir ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314666 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/http.conf.sample | 5 +++++ configs/sip.conf.sample | 10 ++++++++++ configs/skinny.conf.sample | 9 +++++++++ 3 files changed, 24 insertions(+) (limited to 'configs') 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/sip.conf.sample b/configs/sip.conf.sample index 5c8fa4de9..1726299d3 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 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 -- cgit v1.2.3 From aed2cc7cd148eb18135cbc97539841681d047356 Mon Sep 17 00:00:00 2001 From: lmadsen Date: Thu, 5 May 2011 12:28:40 +0000 Subject: Merged revisions 317058 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317058 | lmadsen | 2011-05-05 08:27:56 -0400 (Thu, 05 May 2011) | 7 lines Remove unused directory and clear up some documentation. (closes issue #19193) Reported by: bchia Patches: cel-csv.diff uploaded by lathama (license 1028) Tested by: lathama, Marquis42 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317059 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/cel.conf.sample | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'configs') 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] -- cgit v1.2.3 From b1614a0ef555ea0a00578050ecf75c943e28d3a6 Mon Sep 17 00:00:00 2001 From: russell Date: Thu, 5 May 2011 23:08:05 +0000 Subject: Add CEL extra field to cel_pgsql. (closes issue #18462) Reported by: joscas Patches: bug_18462.diff uploaded by snuffy (license 35) cel_pgsql.conf.sample.issue18462.patch uploaded by joscas (license 1180) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317482 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/cel_pgsql.conf.sample | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'configs') 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 -- cgit v1.2.3 From 8b669b808bad397a6be9fae517e05271a5cb1d5a Mon Sep 17 00:00:00 2001 From: mnicholson Date: Fri, 6 May 2011 19:19:56 +0000 Subject: Updated the sample pbx_lua config file to reflect autoservice changes. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317818 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/extensions.lua.sample | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) (limited to 'configs') 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) -- cgit v1.2.3 From 0e5dc27d665e206e4672722e9205c4f55d081f65 Mon Sep 17 00:00:00 2001 From: jrose Date: Mon, 9 May 2011 14:21:33 +0000 Subject: Merged revisions 318148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | 4 lines Documenting an observed behavior of features in features.conf. Since parkinglots use an integer for the parkinglot extensions, leading zeros specified in the configuration file are ignored. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@318162 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/features.conf.sample | 3 +++ 1 file changed, 3 insertions(+) (limited to 'configs') 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 -- cgit v1.2.3 From 1aa4733de1ad1d1e260d19b5dce51d9aa7492ab4 Mon Sep 17 00:00:00 2001 From: rmudgett Date: Tue, 17 May 2011 20:13:27 +0000 Subject: Option needed for Q931_IE_TIME_DATE to be optional in CONNECT message. The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG. Add option to specify if and how much of the current time is put in Q931_IE_TIME_DATE. * Send date/time ie never. * Send date/time ie date only. * Send date/time ie date and hour. * Send date/time ie date, hour, and minute. * Send date/time ie date, hour, minute, and second. * Send date/time ie default: Libpri will send date and hhmm only when in NT PTMP mode to support ISDN phones. (closes issue #19221) Reported by: kenner JIRA SWP-3396 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319427 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/chan_dahdi.conf.sample | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'configs') diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample index 604f91708..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 -- cgit v1.2.3 From ef96ee5356196b696ddaac9acfe8e2a264ab88ce Mon Sep 17 00:00:00 2001 From: jrose Date: Fri, 20 May 2011 13:42:15 +0000 Subject: Merged revisions 319938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r319938 | jrose | 2011-05-20 08:28:24 -0500 (Fri, 20 May 2011) | 12 lines Adds legacy_useroption_parsing to address interoperability concerns. With the new option engaged, Asterisk should interpret user fields with useroptions contained within the userfield of the uri by stripping them out of the original message whenever a semicolon is encountered in the userfield string. (closes issue #18344) Reported by: danimal Tested by: jrose Review: https://reviewboard.asterisk.org/r/1223/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@319939 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/sip.conf.sample | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'configs') diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 1726299d3..179678a39 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -432,6 +432,14 @@ 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 -- cgit v1.2.3 From c321368c4834325a3fedae4e54134d8e5182d339 Mon Sep 17 00:00:00 2001 From: russell Date: Wed, 1 Jun 2011 21:31:40 +0000 Subject: Support routing text messages outside of a call. Asterisk now has protocol independent support for processing text messages outside of a call. Messages are routed through the Asterisk dialplan. SIP MESSAGE and XMPP are currently supported. There are options in sip.conf and jabber.conf that enable these features. There is a new application, MessageSend(). There are two new functions, MESSAGE() and MESSAGE_DATA(). Documentation will be available on the project wiki, wiki.asterisk.org. Thanks to Terry Wilson for the assistance with development and to David Vossel for helping with some additional testing. Review: https://reviewboard.asterisk.org/r/1042/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321546 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/jabber.conf.sample | 3 +++ configs/sip.conf.sample | 10 ++++++++++ 2 files changed, 13 insertions(+) (limited to 'configs') 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/sip.conf.sample b/configs/sip.conf.sample index 179678a39..49277d64f 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -385,6 +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 -- cgit v1.2.3 From bde62216a95a39a3189fd919194d48beefe92818 Mon Sep 17 00:00:00 2001 From: lmadsen Date: Fri, 3 Jun 2011 13:18:21 +0000 Subject: Merged revisions 321685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r321685 | lmadsen | 2011-06-03 08:17:50 -0500 (Fri, 03 Jun 2011) | 5 lines Also document the 'queue-minute' option. (closes issue #19386) Reported by: juanmol ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@321689 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/queues.conf.sample | 2 ++ 1 file changed, 2 insertions(+) (limited to 'configs') 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.") -- cgit v1.2.3 From 21edfd30886c48626d7f4817d98f9d3c08a49b2e Mon Sep 17 00:00:00 2001 From: pabelanger Date: Tue, 7 Jun 2011 18:01:28 +0000 Subject: Merged revisions 322189 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322189 | pabelanger | 2011-06-07 13:59:13 -0400 (Tue, 07 Jun 2011) | 4 lines Use correct syntax for 'sip notify snom-reboot' (closes issue ASTERISK-17915) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@322190 f38db490-d61c-443f-a65b-d21fe96a405b --- configs/sip_notify.conf.sample | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'configs') 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 -- cgit v1.2.3