Asterisk standard extensions Chinese

Asterisk standard extensions 中文版

Standard extensions

底下的部份中文翻譯尚未完善，待英文較好的人來補足

Page Contents

Explanations
Impact on CDR

Standard Priorities
See also

a: Called 當使用者按 '*' 號，在語音信箱提示期間
h: 掛斷(Hangup)分機
i: 無效分機
o: Operator extension, used for operator exit by pressing zero in voicemail
s: 初始分機 in context
t: 超時分機
T: AbsoluteTimeout() extension
failed: 用於 auto-dial out 撥打失敗 (that had context, priority and extension specified)
fax: 用於傳真偵測on Zap channels
talk: used in conjuction with BackgroundDetect

Explanations

s: 開始. 用於 dialplans 內的第一個分機號，當一通電話進入了此context中，卻不帶任何分機號的資訊。想像一下沒有DID(直接撥入號碼)的電話線，電話進來，我們只知鈴聲響了，其它的什麼都不知道。即使你知道有來電顯示，你仍須將它放至 start 分機。You can also think about s as a place to place part of the dialplan that you don't want callers to get back to unless they have passed through other functions.
t: 逾時. Used for when calls have been inactive after a prompt was played. Also used to hang up a line that has been idle.
T: 絕對逾時. Used for calls that have been hung up due to an AbsoluteTimeout() being reached. For example useful to play a notification with Playback().
h: 掛斷。通常用來清除一通電話，可用於在掛斷通話之前撥放一些訊息。一般用於電話卡系統，記錄一通電話結束時的費用。若通話被停泊了(parked)，則不會執行 h 分機號。More...
i: 用於撥號至context內未知的分機號，或在IVR選單時，輸入未知的號碼。

Impact on CDR

Using these extensions may change the content of your CDR billing records in a way you do not want: In particular the dst field will not contain the dialed number anymore! As a workaround you could look into Dial() with the 'g' option ("go on in context") and consider using ResetCDR(w) and / or NoCdr().

Standard Priorities

hint: hint 優先值(priority) 與分機和 Asterisk 頻道有關，目地是為了對應頻道狀態 to 分機狀態。

在 asterisk 裡，頻道(設備)可以有多種狀態 (unavailable, in-use, busy, ringing....等等)但一個分機只能標記一個序號給 app 使用。然而，當溝通時頻道狀態 to 外部裝置時，如 receptionist console，你無法使用 asterisk 內部頻道的名稱，必須使用外部識別資源名稱，一般來說，就是分機號碼。

然而，一個設備的狀態與分機的狀態他們之間的聯繫(狀態的接收與通知)，都是透過 SIP channel(頻道)(訂閱/通知的機制經由文件RFC-3265實作）去點亮 SIP 話機上的訊號燈。
This is supported in SNOM phones (see also) with their programmable keys set to type "destination", as well as in Polycom (500/600), Aastra ( 480i, 9133i ), and Sayson phones.

For hints/BLF to work as expected in Asterisk 1.4 you need to ensure that the following configuration items are set in sip.conf:

Asterisk sip call-limit must be set for every extension. Usually set this to a large arbitrary number such as 100. (or 1 if you don't want another call to be received when the extension is on the phone)
Asterisk sip notifyringing should be set to "yes".
If you're using RealTime, ensure you have both rtcachefriends and rtupdate set to "yes".
limitonpeers must be set to "yes" - this is the most likely culprit since it's missing from the 1.4 default config files

Note: Up to and including Asterisk 1.4 Local channels do not support state information and can therefore not be watched with hint. Asterisk 1.6 changes this.

Asterisk 1.6.1.x: It is now possible to specify a pattern match as a hint. Once a phone subscribes to something that matches the pattern a hint will be created using the contents and variables evaluated.

Privacy considerations: In sip.conf you can define a subscribecontext= value that determines in which context Asterisk should search for the matching extension when a subscribe request is received from the phone; however, if the extension doesn't exist in that context Asterisk is going to look for it in the default context! In other words: Everyone can subscribe to a "hinted" extension that is defined in the default context. By the way, specifying an empty subscribecontext is also fine if the phone should not at all subscribe to _any_ context.

Likewise bug/patch 5515 (post Asterisk 1.2.0) adds devstate support also for MGCP (so far SIP, IAX and ZAP are supported; "show channeltypes" tell you which channels in your Asterisk support device status notification). Question: Does this patch only show a device which is unavailable (e.g. disconnected), or does it also show "busy"? Answer: Also "busy" (in use).

Also chan_capi-cm v0.6.2 and later comes with basic hint support. It appears, however, that the dynamic naming of CAPI channels that includes the called number makes monitoring of a CAPI line for outgoing calls practically impossible - at least for now.

Useful CLI commands for debugging are "SIP show subscriptions", "show hints", "show channeltypes" and "SIP show inuse".

Related CLI commands in asterisk 1.4: "sip show subscriptions", "sip show inuse", "core show hints", "core show channeltypes"
Related options in sip.conf: notifyringing, notifyhold, limitonpeers, subscribecontext, busy-level(1.6.x only), call-limit

app_devstate

The 3rd party Bristuff patches come with app_devstate that permits state manipulation through the dialplan.

New: While Asterisk 1.6 will include func_devstate natively there is now also a backport available for 1.4. This is quite similar to app_devstate as part of the bristuff patches.

At this little web page a small PHP utility is available that allows to transfer BLF/hint presence information across several Asterisk servers.
It appears that bristuff patches introduced a solution for the same aim in 0.3.0-PRE-1t with res_esel (Extension State Export Logic). In the meanwhile Russel Bryant of Digium is looking at a similar, native feature for Asterisk 1.6 (or 1.8) via res_ais using AIS

New in Asterisk 1.4

Deny subscriptions

If you don’t want you Asterisk server to accept any registrations, you can simply add “allowsubscribe=no” to your sip.conf and stop reading here. This can be done on a per-account basis, as well as in the [general] configuration section.

Display hold states

Asterisk can now display hold state in both XML-RPID (SIMPLE) mode and dialog-info subscriptions. On most phones, this will be shown as a slowly blinking lamp. To enable this, add “notifyhold=yes” to the sip.conf [general] section.

Show inbound and outbound calls

In 1.2, the subscription only applied to the peer part of a type=friend object. This meant that incoming calls from a phone did not turn on the lamp, indicating busy (??). In Asterisk 1.4, you can add “limitonpeers=yes” so that the call limit and counters always will apply on the peer part of a friend. Suddenly, you have blinking lamps for both inbound and outbound calls.

Subscribe to parking lots

With the “metermaid” subsystem, you can add a subscription to the state of a parking lot. You specify a hint like “exten => 6600,hint,park:701@parkedcalls” in your diaplan and then subscribe to the extension, in this case 6600. If you want to watch if there’s an active conference, you add “meetme:1234″ to the hint if you want to check conference number 1234.

Continue reading here.

New in Asterisk 1.6

Asterisk 1.6.1.x: It is now possible to specify a pattern match as a hint. Once a phone subscribes to something that matches the pattern a hint will be created using the contents and variables evaluated.

Read about how to transmit state information across Asterisk servers in Asterisk 1.6.

Asterisk 1.6.1.x: The event infrastructure in Asterisk got another big update to help support distributed events. It currently supports distributed device state and distributed Voicemail MWI (Message Waiting Indication). A new module has been merged, res_ais, which facilitates communicating events between servers. It uses the SAForum AIS (Service Availability Forum Application Interface Specification) CLM (Cluster Management) and EVT (Event) services to maintain a cluster of Asterisk servers, and to share events between them. For more information on setting this up, see doc/distributed_devstate.txt.

New in Asterisk 1.8

Event thogh this is a little off-topic it is still related: Distributed device state using XMPP and PubSub

exten => 12345,hint,confbridge:MyConfName ; requires Asterisk 1.8 with ConfBridge()

Examples

Example 1

exten => 200,hint,SIP/phone1 ; this is case sensitive (!) in 1.0.9 and 1.2.0
exten => 200,1,Macro(stdexten,SIP/phone1)

Example 2

If you want to monitor the state of multiple phones using one speeddial, you can do so:

exten => 200,hint,SIP/201&SIP/202&SIP/203

Asterisk provides a syntax for allowing more than one channel to be mapped to any particular extension with the hint system.

Example 3: Callpark indication (Asterisk 1.4 and newer):

include => parkedcalls
exten => 701,1,ParkedCall(701)
exten => 701,hint,park:701@parkedcalls

Example 4: Dynamic hints in Asterisk 1.6.1

Q: Is it possible to have something like

exten => _XXX,hint,SIP/${EXTEN}

in extensions.conf so that I don't have to add hint for each sip peer I create?

A:: Only in 1.6.1 and later. The hints will grow, as phones subscribe to them, one entry per hint, automatically.
When the phone unsubscribes the hint will continue to exist until a reload is performed. This is how dynamic hints work - they are created as needed and disappear on reload, if they are no longer being watched.