Asterisk channel - Revision history

Pchero: /* Standard channel variables */

2017-04-06T14:16:57Z

Standard channel variables

← Older revision		Revision as of 14:16, 6 April 2017
Line 139:		Line 139:

	So now only two channels exists; the one from SIP/201 to asterisk and one from asterisk to SIP/202, and they are directly bridged together.		So now only two channels exists; the one from SIP/201 to asterisk and one from asterisk to SIP/202, and they are directly bridged together.

	~~== Standard channel variables ==~~
	~~"*" 가 붙은 변수들은 dialplan 에서 write 가 불가능하고 오직 read 만 가능한 변수들이다. 만약 write 를 할 경우, 경고나 에러 메시지 없이 무시된다.~~

	~~=== Variables present in Asterisk 1.8 and forward ===~~
	~~Asterisk-1.8 부터 계속 사용되고 있는 channel variable 내용 정리.~~
	* ${CDR(accountcode)}* : Account code(if specified).
	* ${BLINDTRANSFER} : The name of the channel on the other side of a blind transfer.
	* ${BRIDGEPEER} : Bridged peer.
	* ${BRIDGEPVTCALLID} : Bridged peer PVT call ID(SIP call ID if a SIP call).
	* ${CALLERID(ani)}* : Caller ANI(PRI channels).
	* ${CALLERID(ani2)}* : ANI2(info digits) also called Originating line information or OLI.
	* ${CALLERID(all)} : Caller ID.
	* ${CALLERID(dnid)}* : Dialed Number Identifer.
	* ${CALLERID(name)}* : Caller ID Name only.
	* ${CALLERID(rdnis)}* : Redirected Dial Number ID Service.
	* ${CALLINGANI2}* : Caller ANI2(PRI channels).
	* ${CALLINGPRES}* : Caller ID presentation for incoming calls(PRI channels).
	* ${CALLINGTNS}* : Transit network Selector(PRI channels).
	* ${CALLINGTON}* : Caller Type of Number(PRI channels).
	* ${CHANNEL}* : Current channel name.
	* ${CONTEXT}* : Current context.
	* ${DATETIME}* : Current date time in the format: DDMMYYYY-HH:MM:SS(Deprecated; use ${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}).
	* ${DB_RESULT} : Result value of DB_EXISTS() dial plan function.
	* ${EPOCH}* : Current unix style epoch.
	* ${EXTEN}* : Current extension.
	* ${ENV(VAR)} : Environmental variable VAR.
	* ${GOTO_ON_BLINDXFR} : Transfer to the specified context/extension/priority after a blind transfer(use ^ characters in place of \| to separate context/extension/prioirty when setting this variable from the dialplan).
	* ${HANGUPCAUSE}* : Asterisk cause of hangup(inbound/outbound).
	* ${HINT}* : Channel hints for this extension.
	* ${HINTNAME}* : Suggested Caller ID name for this extension.
	* ${INVALID_EXTEN} : The invalid called extension(used in the "i" extension).
	* ${LANGUAGE}* : Current language(Deprecated; use ${CHANNEL(language)}).
	* ${LEN(VAR)} : String length of VAR(integer).
	* ${PRIORITY}* : Current language(Deprecated; use ${CHANNEL(language)}).
	* ${PRIREDIRECTREASON} : Reason for redirect on PRI, if a call was directed.
	* ${TIMESTAMP}* : Current date time in the format: YYYYMMDD-HHMMSS(Deprecated; use ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}).
	* ${TRANSFER_CONTEXT} : Context for transferred calls.
	* ${FORWARD_CONTEXT} : Context for forwarded calls.
	* ${DYNAMIC_PEERNAME} : The name of the channel on the other side when a dynamic feature is used.
	* ${DYNAMIC_FEATURENAME} : The name of the last triggered dynamic feature.
	* ${UNIQUEID}* : Current call unique identifier.
	* ${SYSTEMNAME}* : Value of the systemname option of asterisk.conf.
	* ${ENTITYID}* : Global Entity ID set automatically, or from asterisk.conf.

	~~=== Variables present in Asterisk 11 and forward ===~~
	~~Asterisk-11 부터 계속 사용되고 있는 channel variable 내용 정리.~~

	* ${AGIEXITONHANGUP} : Set to 1 to force the behavior of a call to AGI to behave as it did in 1.4, where the AGI script would exit immediately on detecting a channel hangup.
	* ${CALENDAR_SUCCESS} * : Status of the CALENDAR_WRITE function. Set to 1 if the function completed successfully; 0 otherwise.
	* ${SIP_RECVADDR} * : The address a SIP MESSAGE request was received from.
	* ${VOICEMAIL_PLAYBACKSTATUS} * : Status of the VoiceMailPlayMsg application. SUCCESS if the voicemail was played back successfully, FAILED otherwise.

	== See also ==		== See also ==

Pchero: /* Variables present in Asterisk 1.8 and forward */

2017-02-10T15:03:23Z

Variables present in Asterisk 1.8 and forward

@@ Line 146: / Line 146: @@
 Asterisk-1.8 부터 계속 사용되고 있는 channel variable 내용 정리.
 * ${CDR(accountcode)}* : Account code(if specified).
 === Variables present in Asterisk 11 and forward ===

Pchero: /* Variables present in Asterisk 1.8 and forward */

2017-02-10T14:45:57Z

Variables present in Asterisk 1.8 and forward

← Older revision		Revision as of 14:45, 10 February 2017
Line 145:		Line 145:
	=== Variables present in Asterisk 1.8 and forward ===		=== Variables present in Asterisk 1.8 and forward ===
	Asterisk-1.8 부터 계속 사용되고 있는 channel variable 내용 정리.		Asterisk-1.8 부터 계속 사용되고 있는 channel variable 내용 정리.
			* ${CDR(accountcode)}* : Account code(if specified).

	=== Variables present in Asterisk 11 and forward ===		=== Variables present in Asterisk 11 and forward ===

Pchero: /* See also */

2017-02-09T11:13:43Z

Pchero: /* Optimization and Zombie call */

2017-01-25T15:21:54Z

Optimization and Zombie call

← Older revision		Revision as of 15:21, 25 January 2017
Line 112:		Line 112:
	- SIP/201 -> [Incoming]		- SIP/201 -> [Incoming]
	- [Incoming] -> [SomeOtherContext]		- [Incoming] -> [SomeOtherContext]
	- [SomeOtherContext -> [DoTheDial]		- [SomeOtherContext] -> [DoTheDial]
	- [DoTheDial] -> SIP/202		- [DoTheDial] -> SIP/202
	</pre>		</pre>

Pchero: /* Local Channel Optimization */

2017-01-25T15:21:27Z

Local Channel Optimization

@@ Line 98: / Line 98: @@
 exten => 4,1,Dial(Local/2@services,,L(60000:45000:15000));
 </pre>
 == See also ==

Pchero: /* See also */

2017-01-16T13:54:04Z

Pchero: /* Local Channel Optimization */

2017-01-16T13:45:20Z

Local Channel Optimization

@@ Line 45: / Line 45: @@
 Figure 3
 : This is what the call would look like when established if you called the Local Channel with "/n". You can see the Local Channels get pushed into bridges with channels they were connected with through app_dial previously.
 == See also ==

Pchero: /* Local Channel */

2017-01-16T12:55:23Z

Local Channel

← Older revision		Revision as of 12:55, 16 January 2017
Line 35:		Line 35:
	This can have some adverse effects when you're expecting information to be available during the call that gets associated with the Local channel. When the Local channel is optimized out of the call path, any Dial() flags, or channel variables associated with the Local channel are also destroyed and are no longer available to Asterisk.		This can have some adverse effects when you're expecting information to be available during the call that gets associated with the Local channel. When the Local channel is optimized out of the call path, any Dial() flags, or channel variables associated with the Local channel are also destroyed and are no longer available to Asterisk.

	~~Diagrams really help to show what is going on~~:		[[File:Asterisk LocalChannels Optimization.png]]

			Figure 1
			: This is a call in an unanswered (ringing) state - from SIP to SIP using Local Channels in between.

			Figure 2
			: By default, after the callee answers this is what the call would look like with the Local Channels optimizing out.

			Figure 3
			: This is what the call would look like when established if you called the Local Channel with "/n". You can see the Local Channels get pushed into bridges with channels they were connected with through app_dial previously.

	== See also ==		== See also ==

Pchero at 12:51, 16 January 2017

2017-01-16T12:51:35Z

← Older revision		Revision as of 12:51, 16 January 2017
Line 27:		Line 27:

	Local channels are used for dialing inward to the system itself, allowing any Asterisk component that can dial to call directly into dialplan. This provides sort of "fake" call that still executes real instructions.		Local channels are used for dialing inward to the system itself, allowing any Asterisk component that can dial to call directly into dialplan. This provides sort of "fake" call that still executes real instructions.

			== Local Channel ==
			=== Local Channel Optimization ===
			==== Default Channel Optimization ====
			By default, the Local channel will try to optimize itself out of the call path. This means that once the Local channel has established the call between the destination and Asterisk, the Local channel will get out of the way and let Asterisk and the end point talk directly, instead of flowing through the Local channel.

			This can have some adverse effects when you're expecting information to be available during the call that gets associated with the Local channel. When the Local channel is optimized out of the call path, any Dial() flags, or channel variables associated with the Local channel are also destroyed and are no longer available to Asterisk.

			Diagrams really help to show what is going on:


	== See also ==		== See also ==
	* https://wiki.asterisk.org/wiki/display/AST/Channels - Channels		* https://wiki.asterisk.org/wiki/display/AST/Channels - Channels
			* https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Optimization - Local Channel Optimization

	[[category:Asterisk]]		[[category:Asterisk]]

← Older revision		Revision as of 11:13, 9 February 2017
Line 139:		Line 139:

	So now only two channels exists; the one from SIP/201 to asterisk and one from asterisk to SIP/202, and they are directly bridged together.		So now only two channels exists; the one from SIP/201 to asterisk and one from asterisk to SIP/202, and they are directly bridged together.

			== Standard channel variables ==
			"*" 가 붙은 변수들은 dialplan 에서 write 가 불가능하고 오직 read 만 가능한 변수들이다. 만약 write 를 할 경우, 경고나 에러 메시지 없이 무시된다.

			=== Variables present in Asterisk 1.8 and forward ===
			Asterisk-1.8 부터 계속 사용되고 있는 channel variable 내용 정리.

			=== Variables present in Asterisk 11 and forward ===
			Asterisk-11 부터 계속 사용되고 있는 channel variable 내용 정리.

			* ${AGIEXITONHANGUP} : Set to 1 to force the behavior of a call to AGI to behave as it did in 1.4, where the AGI script would exit immediately on detecting a channel hangup.
			* ${CALENDAR_SUCCESS} * : Status of the CALENDAR_WRITE function. Set to 1 if the function completed successfully; 0 otherwise.
			* ${SIP_RECVADDR} * : The address a SIP MESSAGE request was received from.
			* ${VOICEMAIL_PLAYBACKSTATUS} * : Status of the VoiceMailPlayMsg application. SUCCESS if the voicemail was played back successfully, FAILED otherwise.

	== See also ==		== See also ==
	* https://wiki.asterisk.org/wiki/display/AST/Channels - Channels		* https://wiki.asterisk.org/wiki/display/AST/Channels - Channels
	* https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Optimization - Local Channel Optimization		* https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Optimization - Local Channel Optimization
			* https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables

	== Reference ==		== Reference ==

← Older revision		Revision as of 13:54, 16 January 2017
Line 102:		Line 102:
	* https://wiki.asterisk.org/wiki/display/AST/Channels - Channels		* https://wiki.asterisk.org/wiki/display/AST/Channels - Channels
	* https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Optimization - Local Channel Optimization		* https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Optimization - Local Channel Optimization

			== Reference ==
			<references />

	[[category:Asterisk]]		[[category:Asterisk]]