Asterisk

From Luke Jackson

Revision as of 22:40, 1 December 2006 (edit) Ljackson (Talk | contribs) (→Tutorials) ← Previous diff		Current revision (22:03, 20 December 2007) (edit) Ljackson (Talk | contribs) (→How do I enable Attended Transfer?)
Line 17:		Line 17:
	==Download==		==Download==
		+
		+	* http://asterisk.org/downloads
	==Hardware==		==Hardware==
		+
		+	* Digium
		+	* Sangoma
	==Software==		==Software==
-	==Tutorials==	+	* FreePBX
		+	* ThirdLane
		+
		+	== Tutorials ==
	* [[Asterisk Echo]]		* [[Asterisk Echo]]
		+	* [[Asterisk Install]]
		+	* [[Asterisk Firewall]]
		+	* [[Asterisk Front End]]
		+	* [[Asterisk Logs]]
		+	* [[Asterisk Queue]]
		+	* [[Asterisk Sipgate]]
		+	* [[Server Backup#Asterisk, MySQL, FreePBX]]
		+
		+	== FAQs ==
		+
		+	=== MeetMe: "That is not a valid conference number." ===
		+
		+	Ensure your meetme.conf file uses '''","''' commas and not '''"|"''' pipes.
		+
		+	meetme.conf:
		+
		+	conf => 1995,1234
		+
		+	=== How do I use "Console commands" (transfer, dial, answer, hangup) from the cli? ===
		+
		+	You'll need chan_oss for that. Chan_oss connects the sound card to the outgoing line.
		+	By default it is disabled to include the module chan_oss, you will need to edit a section of the modules.conf file to match the sample below:
		+
		+	<pre>
		+	; Load either OSS or ALSA, not both
		+	; By default, load OSS only (automatically) and do not load ALSA
		+	;
		+	noload => chan_alsa.so
		+	;noload => chan_oss.so
		+	</pre>
		+
		+	=== I configured a new IVR and clicked the check box that states "Enable Directory". When I call in and type # I get "I'm sorry, that's not a valid extension." What did I do wrong? ===
		+
		+	You must install '''Information Services''', It provides a number of applications accessible by feature codes:
		+
		+	* Company directory
		+	* Call Trace (last call information)
		+	* Echo Test
		+	* Speaking Clock
		+	* Speak Current Extension Number
		+
		+	More info: [http://aussievoip.com.au/wiki/freePBX-Features http://aussievoip.com.au/wiki/freePBX-Features]
		+
		+	=== How do I enable Attended Transfer? ===
		+
		+	Update your features.conf file to reflect the settings below:
		+
		+	<pre>
		+	[featuremap]
		+	blindxfer => # ; Blind Transfer
		+	disconnect => *9 ; Disconnect Call
		+	automon => *0 ; One Touch Record
		+	atxfer => * ; Attended Xfer
		+	</pre>
		+
		+	=== Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) ===
		+
		+	Check to ensure that the user you are attempting to connect as has full rights to /var/run/asterisk.
		+
		+	Change /etc/asterisk.conf:
		+
		+	astrundir => /var/run
-	==FAQs==	+	to
-	'''How do I dial from the CLI?'''	+	astrundir => /var/run/asterisk
-	You'll need chan_oss for that. chan_oss provides a "dial" command form the CLI that literally connects the soundcard (via chan_oss) to the outgoing line. By including the module chan_oss, you'll find a new "dial" command on the CLI.
	[[Category:VoIP]]		[[Category:VoIP]]
	[[Category:Linux]]		[[Category:Linux]]
	[[Category:Mac OS X]]		[[Category:Mac OS X]]

---

Current revision

To learn more about Asterisk and its partners you can visit their website at Asterisk.org.

Contents [hide] 1 Introduction 2 Download 3 Hardware 4 Software 5 Tutorials 6 FAQs 6.1 MeetMe: "That is not a valid conference number." 6.2 How do I use "Console commands" (transfer, dial, answer, hangup) from the cli? 6.3 I configured a new IVR and clicked the check box that states "Enable Directory". When I call in and type # I get "I'm sorry, that's not a valid extension." What did I do wrong? 6.4 How do I enable Attended Transfer? 6.5 Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)

Introduction

Asterisk is an open source PBX system then runs on the Linux operating system.

I support Asterisk because of its price benefits and ease of expansion.

Some key features of Asterisk are listed below.

• Voicemail to eMail
• Softphone (Computer Based Phone) support
• Analog Phone Support (PSTN)
• ISDN PRI support for DID based routing
• VoIP support

Download

• http://asterisk.org/downloads

Hardware

• Digium
• Sangoma

Software

• FreePBX
• ThirdLane

Tutorials

• Asterisk Echo
• Asterisk Install
• Asterisk Firewall
• Asterisk Front End
• Asterisk Logs
• Asterisk Queue
• Asterisk Sipgate
• Server Backup#Asterisk, MySQL, FreePBX

FAQs

MeetMe: "That is not a valid conference number."

Ensure your meetme.conf file uses "," commas and not "|" pipes.

meetme.conf:

conf => 1995,1234

How do I use "Console commands" (transfer, dial, answer, hangup) from the cli?

You'll need chan_oss for that. Chan_oss connects the sound card to the outgoing line. By default it is disabled to include the module chan_oss, you will need to edit a section of the modules.conf file to match the sample below:

; Load either OSS or ALSA, not both
; By default, load OSS only (automatically) and do not load ALSA
;
noload => chan_alsa.so
;noload => chan_oss.so

I configured a new IVR and clicked the check box that states "Enable Directory". When I call in and type # I get "I'm sorry, that's not a valid extension." What did I do wrong?

You must install Information Services, It provides a number of applications accessible by feature codes:

• Company directory
• Call Trace (last call information)
• Echo Test
• Speaking Clock
• Speak Current Extension Number

More info: http://aussievoip.com.au/wiki/freePBX-Features

How do I enable Attended Transfer?

Update your features.conf file to reflect the settings below:

[featuremap]
blindxfer => #                  ; Blind Transfer
disconnect => *9                ; Disconnect Call
automon => *0                   ; One Touch Record
atxfer => *                     ; Attended Xfer

Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)

Check to ensure that the user you are attempting to connect as has full rights to /var/run/asterisk.

Change /etc/asterisk.conf:

astrundir => /var/run

to

astrundir => /var/run/asterisk