BreadCrumbs: Asterisk

Asterisk

From Luke Jackson

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(FAQs)
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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

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

Hardware

  • Digium
  • Sangoma

Software

  • FreePBX
  • ThirdLane

Tutorials

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
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