BreadCrumbs: Asterisk

Asterisk

From Luke Jackson

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(→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==+== FAQs ==
-'''How do I dial from the CLI?'''+=== MeetMe: "That is not a valid conference number." ===
-You'll need chan_oss for that. chan_oss provides a "dial" command form the CLI that literally connects the sound card (via chan_oss) to the outgoing line. By including the module chan_oss, you'll find a new "dial" command on the CLI.+Ensure your meetme.conf file uses '''","''' commas and not '''"|"''' pipes.
-'''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?'''+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: You must install '''Information Services''', It provides a number of applications accessible by feature codes:
Line 42: Line 74:
* Speak Current Extension Number * Speak Current Extension Number
-More info: [http://aussievoip.com.au/wiki/freePBX-Features]+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
 + 
 +to
 + 
 + astrundir => /var/run/asterisk
 + 
[[Category:VoIP]] [[Category:VoIP]]
[[Category:Linux]] [[Category:Linux]]
[[Category:Mac OS X]] [[Category:Mac OS X]]

Current revision

Asterisk Logo

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

Contents

[edit]

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
[edit]

Download

[edit]

Hardware

  • Digium
  • Sangoma
[edit]

Software

  • FreePBX
  • ThirdLane
[edit]

Tutorials

[edit]

FAQs

[edit]

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

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

meetme.conf: 

conf => 1995,1234
[edit]

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
[edit]

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

[edit]

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
[edit]

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
Retrieved from "http://wiki.ljackson.us/Asterisk"
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