diff --git a/voice-client/files/etc/asterisk_templates/asterisk.conf.TEMPLATE b/voice-client/files/etc/asterisk_templates/asterisk.conf.TEMPLATE index 46f33037f..c6193ab21 100644 --- a/voice-client/files/etc/asterisk_templates/asterisk.conf.TEMPLATE +++ b/voice-client/files/etc/asterisk_templates/asterisk.conf.TEMPLATE @@ -1,14 +1,15 @@ -[directories] +[directories](!) astetcdir => /etc/asterisk astmoddir => /usr/lib/asterisk/modules -astvarlibdir => /usr/lib/asterisk -astdbdir => /usr/lib/asterisk -astkeydir => /usr/lib/asterisk -astdatadir => /usr/lib/asterisk -astagidir => /usr/lib/asterisk/agi-bin +astvarlibdir => /var/lib/asterisk +astdbdir => /var/lib/asterisk +astkeydir => /var/lib/asterisk +astdatadir => /usr/share/asterisk +astagidir => /usr/share/asterisk/agi-bin astspooldir => /var/spool/asterisk astrundir => /var/run/asterisk astlogdir => /var/log/asterisk +astsbindir => /usr/sbin [options] ;verbose = 3 @@ -17,7 +18,7 @@ astlogdir => /var/log/asterisk ;nofork = yes ; Same as -f at startup. ;quiet = yes ; Same as -q at startup. ;timestamp = yes ; Same as -T at startup. -execincludes = yes ; Support #exec in config files. +;execincludes = yes ; Support #exec in config files. ;console = yes ; Run as console (same as -c at startup). ;highpriority = yes ; Run realtime priority (same as -p at ; startup). @@ -27,12 +28,14 @@ execincludes = yes ; Support #exec in config files. ;dontwarn = yes ; Disable some warnings. ;dumpcore = yes ; Dump core on crash (same as -g at startup). ;languageprefix = yes ; Use the new sound prefix path syntax. -;internal_timing = yes ;systemname = my_system_name ; Prefix uniqueid with a system name for ; Global uniqueness issues. ;autosystemname = yes ; Automatically set systemname to hostname, ; uses 'localhost' on failure, or systemname if ; set. +;mindtmfduration = 80 ; Set minimum DTMF duration in ms (default 80 ms) + ; If we get shorter DTMF messages, these will be + ; changed to the minimum duration ;maxcalls = 10 ; Maximum amount of calls allowed. ;maxload = 0.9 ; Asterisk stops accepting new calls if the ; load average exceed this limit. @@ -73,7 +76,35 @@ documentation_language = en_US ; Set the language you want documentation ;lockconfdir = no ; Protect the directory containing the ; configuration files (/etc/asterisk) with a ; lock. -mindtmfduration=60 +;stdexten = gosub ; How to invoke the extensions.conf stdexten. + ; macro - Invoke the stdexten using a macro as + ; done by legacy Asterisk versions. + ; gosub - Invoke the stdexten using a gosub as + ; documented in extensions.conf.sample. + ; Default gosub. +;live_dangerously = no ; Enable the execution of 'dangerous' dialplan + ; functions from external sources (AMI, + ; etc.) These functions (such as SHELL) are + ; considered dangerous because they can allow + ; privilege escalation. + ; Default no +;entityid=00:11:22:33:44:55 ; Entity ID. + ; This is in the form of a MAC address. + ; It should be universally unique. + ; It must be unique between servers communicating + ; with a protocol that uses this value. + ; This is currently is used by DUNDi and + ; Exchanging Device and Mailbox State + ; using protocols: XMPP, Corosync and PJSIP. +;rtp_pt_dynamic = 96 ; Normally the Dynamic RTP Payload Type numbers + ; are 96-127, which allow 32 formats. When you + ; use more and receive the message "No Dynamic + ; RTP mapping available", extend the dynamic + ; range by going for 35 (or 0) instead of 96. + ; This allows 29 (or 64) more formats. 96 is the + ; default because any number below might be + ; rejected by a remote implementation; although + ; no such broken implementation is known, yet. ; Changing the following lines may compromise your security. ;[files] @@ -81,8 +112,3 @@ mindtmfduration=60 ;astctlowner = root ;astctlgroup = apache ;astctl = asterisk.ctl - -[compat] -pbx_realtime=1.6 -res_agi=1.6 -app_set=1.6 diff --git a/voice-client/files/etc/asterisk_templates/modules.conf.TEMPLATE b/voice-client/files/etc/asterisk_templates/modules.conf.TEMPLATE index 0296bc573..38e33e1f4 100644 --- a/voice-client/files/etc/asterisk_templates/modules.conf.TEMPLATE +++ b/voice-client/files/etc/asterisk_templates/modules.conf.TEMPLATE @@ -1,36 +1,45 @@ +; +; Asterisk configuration file +; +; Module Loader configuration file +; + [modules] autoload=yes -load => chan_brcm.so -load => res_jabber.so ; Used for Gtalk -load => res_clioriginate.so ; originate calls from commandline -load => res_rtp_asterisk.so ; rtp "engine" is now a loadable module in asterisk 1.8 -load => pbx_config.so ; Text Extension Configuration Requires N/A -load => func_callerid.so ; Gets or sets Caller*ID data on the channel. - Requires ? -load => func_channel.so -load => func_logic.so ; Logic functions (if, etc.) -load => func_strings.so ; string manipulation functions -load => cdr_manager.so ; Asterisk Call Manager CDR Backend - Requires N/A -load => chan_local.so ; Show status of local channels- Requires N/A -load => chan_gtalk.so ; Use gtalk -load => chan_sip.so ; Session Initiation Protocol (SIP) - Requires res_features.so -load => codec_alaw.so ; A-law Coder/Decoder - Requires N/A -load => codec_a_mu.so ; A-law and Mulaw direct Coder/Decoder - Requires N/A -load => codec_gsm.so ; GSM/PCM16 (signed linear) Codec Translat - Requires N/A -load => codec_ulaw.so ; Mu-law Coder/Decoder - Requires N/A -load => format_gsm.so ; Raw GSM data - Requires N/A -load => format_pcm.so ; Raw uLaw 8khz Audio support (PCM) - Requires N/A -load => app_dial.so ; Dialing Application - Requires res_features.so, res_musiconhold.so -load => app_parkandannounce.so ; Call Parking and Announce Application - Requires res_features.so -load => app_playback.so ; Sound File Playback Application - Requires N/A -load => app_system.so ; Execute a system command - Requires N/A -load => app_disa.so ; Direct Inward System Access -load => app_authenticate.so ; Authenticate via pin -load => app_senddtmf.so ; Ability to send DTMF tones on the line. -load => func_cut.so ; To manipulate strings -load => res_musiconhold.so -load => func_db.so +; +; Any modules that need to be loaded before the Asterisk core has been +; initialized (just after the logger has been initialized) can be loaded +; using 'preload'. This will frequently be needed if you wish to map all +; module configuration files into Realtime storage, since the Realtime +; driver will need to be loaded before the modules using those configuration +; files are initialized. +; +; An example of loading ODBC support would be: +;preload => res_odbc.so +;preload => res_config_odbc.so +; +; Uncomment the following if you wish to use the Speech Recognition API +;preload => res_speech.so +; +; If you want Asterisk to fail if a module does not load, then use +; the "require" keyword. Asterisk will exit with a status code of 2 +; if a required module does not load. +; +; require = chan_sip.so +; If you want you can combine with preload +; preload-require = res_odbc.so +; +; If you want, load the GTK console right away. +; noload => pbx_gtkconsole.so +;load => pbx_gtkconsole.so +; +load => res_musiconhold.so +; +; Load one of: chan_oss, alsa, or console (portaudio). +; By default, load chan_oss only (automatically). +; noload => chan_alsa.so +;noload => chan_oss.so noload => chan_console.so -[global] -chan_modem.so=no +;