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Asterisk

Alternatives

OSNameLicenceRemark
LinuxAsterisknow The Digium Asterisk Linux ditribution
LinuxFreeSWITCHMozilla Public License (MPL)
LinuxFusionPBXMozilla Public License (MPL)Based on FreeSWITCH
LinuxKamailioGPL Former SIP Express Router - SER Wiki with deb's
LinuxVodiaPropreiataryFormer: Pbxnsip
WindowsVoIPswitch
Windows3CXPropreiatary
CloudsipXecsAffero General Public License
Voiceserv Webserver is up, no site 160709 Voiceserve on Wikipedia

Main Asterisk website
Asterisk Exported Documentation
Asterisk configuration
Asterisk guru
The asterisk book. For version 1.4-1.6. Minimal setup tutorial

We installed asterisk on a Raspberry Pi 2B running Raspbian 7.11

Install

In aptitude select

  • asterisk
  • asterisk-dahdi. DAHDI (Digium/Asterisk Hardware Device Interface is the open source device interface technology used to control Digium and other legacy telephony interface cards. So it might be omitted if you do not use those cards, not tested)
  • libpri1.4
  • Enter the ITU-T telephone code ( 31 for The Netherlands )
  • Check if there are any upgradeble packages, if so upgrade them
  • Run asterisk -v . We have version 1.8.13.1~dfsg1-3+deb7u4. This is a manual

Configure

  • cd /etc/asterisk
  • cp extensions.conf extensions.conf.org
  • rm extensions.conf
  • vi extensions.conf
    • Result zie Pi
    • chown asterisk:asterisk extensions.conf
  • cp sip.conf sip.conf.org
  • rm sip.conf
  • vi sip.conf
    • Result zie Pi
    • chown asterisk:asterisk sip.conf
  • NOTE: res_ldap.conf should be chmod 600 because it contains the plain-text LDAP

password to an account with WRITE access to the asterisk configuration.

Errors

  • No such extension in <extension number> context 'default'
    • The number is probably under different headers in sip.conf and extensions.conf
  • NOTICE[29754]: cel_tds.c:452 tds_load_module: cel_tds has no global category, nothing to configure.
    • Remove the “;” from “;[global]” in file cel_tds.conf and the notice is gone
  • ERROR[29754]: res_config_ldap.c:1658 parse_config: No directory URL or host found.
  • ERROR[29754]: res_config_ldap.c:1541 load_module: Cannot load LDAP RealTime driver.
    • Remove the “;” in front of host en port. Let host point to your Asterisk server
      • host=192.168.1.<ASterisk> ; LDAP host
      • port=389
    • Result: One error gone. How to disable LDAP if you do not need it?

Configure a sip phone or softphone with sip

  • SNTP Some SNTP server
  • Fixed IP:
  • Phone IP:
  • Host IP:

LinPhone softphone

Use full asterisk commands

From the commandline

  • asterisk -rx “core restart now”
  • asterisk -c
  • asterisk -r
  • asterisk -rgvvvvv

In the asterisk console

agent show
agent show online
cli show permissions
channel request hangup all
console dial 10@default
console dial 10@from-internal
console dial 10@my-phones
console dial 30@pbx-config
console dial 30@users
console hangup 
core stop now; Stop Asterisk and go back to the shell\\
               Asterisk*CLI>\\
               Disconnected from Asterisk server\\
               Executing last minute cleanups\\
               At an other time we got a different output (maybe other verbosity level):\\
core stop now; Beginning asterisk shutdown....\\
               Executing last minute cleanups\\
                 == Destroying musiconhold processes\\
               Asterisk cleanly ending (0).
core show help dialplan
core show channels; Channel              Location             State   Application(Data)\\             
                    0 active channels\\
                    0 active calls\\
                    26 calls processed\\
dial plan reload
dialplan add extension 321,1,Dial,SIP into home
dialplan add extension 321,1,Dial,"SIP/100" into users replace
dialplan add extension 322,1,Dial,SIP into users
dialplan add extension 322,1,Dial,SIP/100 into users replace
dialplan reload
dialplan show users
exit
help
help console
http show status
iax2 provision
load => pbx_config.so
module show like so
module unload res_config_ldap.so
  Unloaded res_config_ldap.so
  *CLI>  LDAP RealTime driver unloaded.
quit
show users
sip reload
sip show mwi
sip show peers
sip set history on; SIP History Recording Enabled (use 'sip show history')
stop
stop now
voicemail show users

Security

Privilege Escalations with Dialplan Functions


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asterisk.txt · Last modified: 15-06-2018 01:36 by wim