Asterisk
From Luke Jackson
Revision as of 20:45, 7 August 2007 (edit) Ljackson (Talk | contribs) (→MeetMe: "That is not a valid conference number.") ← Previous diff |
Current revision (22:03, 20 December 2007) (edit) Ljackson (Talk | contribs) (→How do I enable Attended Transfer?) |
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atxfer => * ; Attended Xfer | atxfer => * ; Attended Xfer | ||
</pre> | </pre> | ||
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+ | === Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) === | ||
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+ | Check to ensure that the user you are attempting to connect as has full rights to /var/run/asterisk. | ||
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+ | Change /etc/asterisk.conf: | ||
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+ | astrundir => /var/run | ||
+ | |||
+ | to | ||
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+ | astrundir => /var/run/asterisk | ||
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[[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.
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
- 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
Categories: VoIP | Linux | Mac OS X