Einrichten von Asterisk 14.x unter CentOS7

Damit der Daemon später beim Hochfahren des Asterisk-Servers automatisch gestartet wird, aktivieren wir den Autostart des Daemon.

 # systemctl enable asterisk.service 

Created symlink from /etc/systemd/system/multi-user.target.wants/asterisk.service to /usr/lib/systemd/system/asterisk.service.

Wollen wir prüfen ob der Autostart des Daemon richtig gesetzt wurde, fragen wir den Status entsprechend wie folgt ab.

 # systemctl is-enabled asterisk.service 

enabled

Der laufende Asterisk-Server stellt zur Interaktion eine eigene Konsole zur Verfügung. Mit der Option -r können wir uns beim Aufruf des Befehls asterisk mit dieser Konsole verbinden.

 # asterisk -r

Asterisk 14.2.1, Copyright (C) 1999 - 2016, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 14.2.1 currently running on vml000177 (pid = 12988)
vml000177*CLI>

Eine Übersicht über den Befehlsumfang den uns die Asterisk-Konsole zur Verfügung stellt, können wir uns mit dem Befehl help anzeigen lassen.

vml000177*CLI> help

!                              -- Execute a shell command
acl show                       -- Show a named ACL or list all named ACLs
agent logoff                   -- Sets an agent offline
agent show all                 -- Show status of all agents
agent show online              -- Show status of online agents
agent show                     -- Show information about an agent
agi dump html                  -- Dumps a list of AGI commands in HTML format
agi exec                       -- Add AGI command to a channel in Async AGI
agi set debug [on|off]         -- Enable/Disable AGI debugging
agi show commands [topic]      -- List AGI commands or specific help
aoc set debug                  -- enable cli debugging of AOC messages
ari mkpasswd                   -- Encrypts a password
ari set debug                  -- Enable/disable debugging of an ARI application
ari show apps                  -- List registered ARI applications
ari show app                   -- Display details of a registered ARI application
ari show status                -- Show ARI settings
ari show users                 -- List ARI users
ari show user                  -- List single ARI user
bridge kick                    -- Kick a channel from a bridge
bridge show all                -- List all bridges
bridge show                    -- Show information about a bridge
bridge technology show         -- List registered bridge technologies
bridge technology {suspend|unsuspend} -- Suspend/unsuspend a bridge technology
cc cancel                      -- Kill a CC transaction
cc report status               -- Reports CC stats
cdr set debug [on|off]         -- Enable debugging in the CDR engine
cdr show active                -- Display active CDRs for channels
cdr show status                -- Display the CDR status
cdr submit                     -- Posts all pending batched CDR data
cel show status                -- Display the CEL status
channel originate              -- Originate a call
channel redirect               -- Redirect a call
channel request hangup         -- Request a hangup on a given channel
cli check permissions          -- Try a permissions config for a user
cli reload permissions         -- Reload CLI permissions config
cli show aliases               -- Show CLI command aliases
cli show permissions           -- Show CLI permissions
confbridge kick                -- Kick participants out of conference bridges.
confbridge list                -- List conference bridges and participants.
confbridge lock                -- Lock a conference.
confbridge mute                -- Mute participants.
confbridge record start        -- Start recording a conference
confbridge record stop         -- Stop recording a conference.
confbridge show menu           -- Show a conference menu
confbridge show menus          -- Show a list of conference menus
confbridge show profile bridge -- Show a conference bridge profile.
confbridge show profile bridges -- Show a list of conference bridge profiles.
confbridge show profile user   -- Show a conference user profile.
confbridge show profile users  -- Show a list of conference user profiles.
confbridge unlock              -- Unlock a conference.
confbridge unmute              -- Unmute participants.
config list                    -- Show all files that have loaded a configuration file
config reload                  -- Force a reload on modules using a particular configuration file
config show help               -- Show configuration help for a module
core abort shutdown            -- Cancel a running shutdown
core clear profile             -- Clear profiling info
core ping taskprocessor        -- Ping a named task processor
core reload                    -- Global reload
core restart gracefully        -- Restart Asterisk gracefully
core restart now               -- Restart Asterisk immediately
core restart when convenient   -- Restart Asterisk at empty call volume
core set debug channel         -- Enable/disable debugging on a channel
core set debug                 -- Set level of debug chattiness
core set verbose               -- Set level of verbose chattiness
core show applications [like|describing] -- Shows registered dialplan applications
core show application          -- Describe a specific dialplan application
core show calls [uptime]       -- Display information on calls
core show channels [concise|verbose|count] -- Display information on channels
core show channel              -- Display information on a specific channel
core show channeltypes         -- List available channel types
core show channeltype          -- Give more details on that channel type
core show codecs [audio|video|image|text] -- Displays a list of registered codecs
core show codec                -- Shows a specific codec
core show config mappings      -- Display config mappings (file names to config engines)
core show file formats         -- Displays file formats
core show functions [like]     -- Shows registered dialplan functions
core show function             -- Describe a specific dialplan function
core show hanguphandlers all   -- Show hangup handlers of all channels
core show hanguphandlers       -- Show hangup handlers of a specified channel
core show help                 -- Display help list, or specific help on a command
core show hints                -- Show dialplan hints
core show hint                 -- Show dialplan hint
core show image formats        -- Displays image formats
core show license              -- Show the license(s) for this copy of Asterisk
core show profile              -- Display profiling info
core show settings             -- Show some core settings
core show sounds               -- Shows available sounds
core show sound                -- Shows details about a specific sound
core show switches             -- Show alternative switches
core show sysinfo              -- Show System Information
core show taskprocessors       -- List instantiated task processors and statistics
core show threads              -- Show running threads
core show translation          -- Display translation matrix
core show uptime [seconds]     -- Show uptime information
core show version              -- Display version info
core show warranty             -- Show the warranty (if any) for this copy of Asterisk
core stop gracefully           -- Gracefully shut down Asterisk
core stop now                  -- Shut down Asterisk immediately
core stop when convenient      -- Shut down Asterisk at empty call volume
core waitfullybooted           -- Wait for Asterisk to be fully booted
data get                       -- Data API get
data show providers            -- Show data providers
database del                   -- Removes database key/value
database deltree               -- Removes database keytree/values
database get                   -- Gets database value
database put                   -- Adds/updates database value
database query                 -- Run a user-specified query on the astdb
database show                  -- Shows database contents
database showkey               -- Shows database contents
devstate change                -- Change a custom device state
devstate list                  -- List currently known custom device states
dialplan add extension         -- Add new extension into context
dialplan add ignorepat         -- Add new ignore pattern
dialplan add include           -- Include context in other context
dialplan debug                 -- Show fast extension pattern matching data structures
dialplan reload                -- Reload extensions and *only* extensions
dialplan remove context        -- Remove a specified context
dialplan remove extension      -- Remove a specified extension
dialplan remove ignorepat      -- Remove ignore pattern from context
dialplan remove include        -- Remove a specified include from context
dialplan save                  -- Save current dialplan into a file
dialplan set chanvar           -- Set a channel variable
dialplan set extenpatternmatchnew false -- Use the Old extension pattern matching algorithm.
dialplan set extenpatternmatchnew true -- Use the New extension pattern matching algorithm.
dialplan set global            -- Set global dialplan variable
dialplan show                  -- Show dialplan
dialplan show chanvar          -- Show channel variables
dialplan show globals          -- Show global dialplan variables
dnsmgr refresh                 -- Performs an immediate refresh
dnsmgr reload                  -- Reloads the DNS manager configuration
dnsmgr status                  -- Display the DNS manager status
dundi flush [stats]            -- Flush DUNDi cache
dundi lookup                   -- Lookup a number in DUNDi
dundi precache                 -- Precache a number in DUNDi
dundi query                    -- Query a DUNDi EID
dundi set debug {on|off}       -- Enable/Disable DUNDi debugging
dundi show cache               -- Show DUNDi cache
dundi show entityid            -- Display Global Entity ID
dundi show hints               -- Show DUNDi hints in the cache
dundi show mappings            -- Show DUNDi mappings
dundi show peers [registered|include|exclude|begin] -- Show defined DUNDi peers
dundi show peer                -- Show info on a specific DUNDi peer
dundi show precache            -- Show DUNDi precache
dundi show requests            -- Show DUNDi requests
dundi show trans               -- Show active DUNDi transactions
dundi store history {on|off}   -- Enable/Disable DUNDi historic records
features show                  -- Lists configured features
file convert                   -- Convert audio file
group show channels            -- Display active channels with group(s)
hangup request                 -- <no description available>
help                           -- <no description available>
http show status               -- Display HTTP server status
iax2 provision                 -- Provision an IAX device
iax2 prune realtime            -- Prune a cached realtime lookup
iax2 reload                    -- Reload IAX configuration
iax2 set debug {on|off|peer}   -- Enable/Disable IAX debugging
iax2 set debug jb {on|off}     -- Enable/Disable IAX jitterbuffer debugging
iax2 set debug trunk {on|off}  -- Enable/Disable IAX trunk debugging
iax2 set mtu                   -- Set the IAX systemwide trunking MTU
iax2 show cache                -- Display IAX cached dialplan
iax2 show callnumber usage     -- Show current entries in IP call number limit table
iax2 show channels             -- List active IAX channels
iax2 show firmware             -- List available IAX firmware
iax2 show netstats             -- List active IAX channel netstats
iax2 show peer                 -- Show details on specific IAX peer
iax2 show peers                -- List defined IAX peers
iax2 show provisioning         -- Display iax provisioning
iax2 show registry             -- Display IAX registration status
iax2 show stats                -- Display IAX statistics
iax2 show threads              -- Display IAX helper thread info
iax2 show users [like]         -- List defined IAX users
iax2 test losspct              -- Set IAX2 incoming frame loss percentage
iax2 unregister                -- Unregister (force expiration) an IAX2 peer from the registry
indication add                 -- Add the given indication to the country
indication remove              -- Remove the given indication from the country
indication show                -- Display a list of all countries/indications
keys init                      -- Initialize RSA key passcodes
keys show                      -- Displays RSA key information
local show channels            -- List status of local channels
logger add channel             -- Adds a new logging channel
logger mute                    -- Toggle logging output to a console
logger reload                  -- Reopens the log files
logger remove channel          -- Removes a logging channel
logger rotate                  -- Rotates and reopens the log files
logger set level {DEBUG|NOTICE|WARNING|ERROR|VERBOSE|DTMF} {on|off} -- Enables/Disables a specific logging level for this console
logger show channels           -- List configured log channels
manager reload                 -- Reload manager configurations
manager set debug [on|off]     -- Show, enable, disable debugging of the manager code
manager show command           -- Show a manager interface command
manager show commands          -- List manager interface commands
manager show connected         -- List connected manager interface users
manager show eventq            -- List manager interface queued events
manager show events            -- List manager interface events
manager show event             -- Show a manager interface event
manager show settings          -- Show manager global settings
manager show users             -- List configured manager users
manager show user              -- Display information on a specific manager user
media cache create             -- Create an item in the media cache
media cache delete             -- Remove an item from the media cache
media cache refresh            -- Refresh an item in the media cache
media cache show all           -- Show all items in the media cache
media cache show               -- Show a single item in the media cache
mixmonitor {start|stop|list}   -- Execute a MixMonitor command
module load                    -- Load a module by name
module reload                  -- Reload configuration for a module
module show [like]             -- List modules and info
module unload                  -- Unload a module by name
moh reload                     -- Reload MusicOnHold
moh show classes               -- List MusicOnHold classes
moh show files                 -- List MusicOnHold file-based classes
no debug channel               -- Disable debugging on channel(s)
originate                      -- <no description available>
parking show                   -- Show a parking lot or a list of all parking lots.
phoneprov show routes          -- Show registered phoneprov http routes
pjsip reload                   -- <no description available>
presencestate change           -- Change a custom presence state
presencestate list             -- List currently know custom presence states
pri intense debug span         -- <no description available>
queue add member               -- Add a channel to a specified queue
queue reload {parameters|members|rules|all} -- Reload queues, members, queue rules, or parameters
queue remove member            -- Removes a channel from a specified queue
queue reset stats              -- Reset statistics for a queue
queue set penalty              -- Set penalty for a channel of a specified queue
queue set ringinuse            -- Set ringinuse for a channel of a specified queue
queue show                     -- Show status of a specified queue
queue show rules               -- Show the rules defined in queuerules.conf
queue {pause|unpause} member   -- Pause or unpause a queue member
realtime destroy               -- Delete a row from a RealTime database
realtime load                  -- Used to print out RealTime variables.
realtime store                 -- Store a new row into a RealTime database
realtime update                -- Used to update RealTime variables.
realtime update2               -- Used to test the RealTime update2 method
reload                         -- <no description available>
rtcp set debug {on|off|ip}     -- Enable/Disable RTCP debugging
rtcp set stats {on|off}        -- Enable/Disable RTCP stats
rtp set debug {on|off|ip}      -- Enable/Disable RTP debugging
say load [new|old]             -- Set or show the say mode
sip notify                     -- Send a notify packet to a SIP peer
sip prune realtime [peer|all]  -- Prune cached Realtime users/peers
sip qualify peer               -- Send an OPTIONS packet to a peer
sip reload                     -- Reload SIP configuration
sip set debug {on|off|ip|peer} -- Enable/Disable SIP debugging
sip set history {on|off}       -- Enable/Disable SIP history
sip show {channels|subscriptions} -- List active SIP channels or subscriptions
sip show channelstats          -- List statistics for active SIP channels
sip show channel               -- Show detailed SIP channel info
sip show domains               -- List our local SIP domains
sip show history               -- Show SIP dialog history
sip show inuse                 -- List all inuse/limits
sip show mwi                   -- Show MWI subscriptions
sip show objects               -- List all SIP object allocations
sip show peers                 -- List defined SIP peers
sip show peer                  -- Show details on specific SIP peer
sip show registry              -- List SIP registration status
sip show sched                 -- Present a report on the status of the scheduler queue
sip show settings              -- Show SIP global settings
sip show tcp                   -- List TCP Connections
sip show users                 -- List defined SIP users
sip show user                  -- Show details on specific SIP user
sip unregister                 -- Unregister (force expiration) a SIP peer from the registry
sorcery memory cache dump      -- Dump all objects within a sorcery memory cache
sorcery memory cache expire    -- Expire a specific object or ALL objects within a sorcery memory cache
sorcery memory cache populate  -- Clear and populate the sorcery memory cache with objects from the backend
sorcery memory cache show      -- Show sorcery memory cache information
sorcery memory cache stale     -- Mark a specific object or ALL objects as stale within a sorcery memory cache
stun set debug {on|off}        -- Enable/Disable STUN debugging
stun show status               -- Show STUN servers and statuses
timing test                    -- Run a timing test
udptl set debug {on|off|ip}    -- Enable/Disable UDPTL debugging
udptl show config              -- Show UDPTL config options
ulimit                         -- Set or show process resource limits
xmldoc dump                    -- Dump the XML docs to the specified file

Eine Übersicht der wesentlichen Einstellungsoptionen können wir uns z.B. mit dem Befehl core show settings anzeigen lassen.

vml000177*CLI> core show settings

PBX Core settings
-----------------
  Version:                     14.2.1
  Build Options:               LOADABLE_MODULES
  Maximum calls:               Not set
  Maximum open file handles:   1024
  Root console verbosity:      0
  Current console verbosity:   0
  Debug level:                 0
  Maximum load average:        0.000000
  Minimum free memory:         0 MB
  Startup time:                19:27:13
  Last reload time:            19:27:13
  System:                      Linux/3.10.0-327.18.2.el7.x86_64 built by mockbuild on x86_64 2016-12-08 22:53:51 UTC
  System name:                 
  Entity ID:                   52:54:00:9a:ac:d3
  PBX UUID:                    a4a776d9-2284-4ef4-a65a-19e838658ef0
  Default language:            en
  Language prefix:             Enabled
  User name and group:         /
  Executable includes:         Disabled
  Transcode via SLIN:          Enabled
  Transmit silence during rec: Disabled
  Generic PLC:                 Enabled
  Min DTMF duration::          80
  RTP dynamic payload types:   96-127

* Subsystems
  -------------
  Manager (AMI):               Disabled
  Web Manager (AMI/HTTP):      Disabled
  Call data records:           Enabled
  Realtime Architecture (ARA): Disabled

* Directories
  -------------
  Configuration file:          /etc/asterisk/asterisk.conf
  Configuration directory:     /etc/asterisk
  Module directory:            /usr/lib64/asterisk/modules
  Spool directory:             /var/spool/asterisk
  Log directory:               /var/log/asterisk
  Run/Sockets directory:       /run/asterisk
  PID file:                    /run/asterisk/asterisk.pid
  VarLib directory:            /usr/share/asterisk
  Data directory:              /usr/share/asterisk
  ASTDB:                       /var/spool/asterisk/astdb
  IAX2 Keys directory:         /usr/share/asterisk/keys
  AGI Scripts directory:       /usr/share/asterisk/agi-bin

Mit der Tastetnkombination STRG und C verlassen wir die Asterisk Konsole wieder.

vml000177*CLI> 

Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups

 # firewall-cmd --get-active-zones

public
  interfaces: eth0 eth2 eth1

 # firewall-cmd --permanent --zone=internal --change-interface=eth1

The interface is under control of NetworkManager, setting zone to 'internal'.
success

 # firewall-cmd --get-active-zones

internal
  interfaces: eth1
public
  interfaces: eth0 eth2

 # firewall-cmd --permanent --zone=internal --add-rich-rule="rule family="ipv4" port protocol="udp" port="4569" accept"

success

 # firewall-cmd --reload

success

 # firewall-cmd --zone=internal --list-all

internal (active)
  target: default
  icmp-block-inversion: no
  interfaces: eth1
  sources: 
  services: dhcpv6-client ssh
  ports: 
  protocols: 
  masquerade: no
  forward-ports: 
  sourceports: 
  icmp-blocks: 
  rich rules: 
	rule family="ipv4" port port="4569" protocol="udp" accept

 # firewall-cmd --get-active-zones

internal
  interfaces: eth1
public
  interfaces: eth0 eth2

 # firewall-cmd --permanent --zone=public --add-rich-rule="rule family="ipv4" port protocol="udp" port="4569" accept"

success

 # firewall-cmd --reload

success

 # firewall-cmd --zone=public --list-all

external
  target: default
  icmp-block-inversion: no
  interfaces: 
  sources: 
  services: ssh
  ports: 
  protocols: 
  masquerade: yes
  forward-ports: 
  sourceports: 
  icmp-blocks: 
  rich rules: 
        rule family="ipv4" port port="4569" protocol="udp" accept

 # asterisk -r

Asterisk 14.2.1, Copyright (C) 1999 - 2016, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 14.2.1 currently running on vml000177 (pid = 13261)
vml000177*CLI>

vml000177*CLI> iax2 show peers

Name/Username    Host                                           Mask                                      Port           Status      Description                     
33/33            10.0.10.122                              (D)  (null)                                    50548          Unmonitored                                 
demo/asterisk    216.207.245.47                           (S)  255.255.255.255                           4569           Unmonitored Demo System At Digium           
2 iax2 peers [0 online, 0 offline, 2 unmonitored]

 # asterisk -r

Asterisk 14.2.1, Copyright (C) 1999 - 2016, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 14.2.1 currently running on vml000177 (pid = 13261)
vml000177*CLI>

vml000177*CLI> dialplan show 

[ Context '__func_periodic_hook_context__' created by 'func_periodic_hook' ]
  'beep' (CID match '') =>  1. Answer()                                   [func_periodic_hook]
                    2. Playback(beep)                             [func_periodic_hook]
  'hook' (CID match '') =>  1. Set(EncodedChannel=${CUT(HOOK_CHANNEL,-,1-2)}) [func_periodic_hook]
                    2. Set(GROUP_NAME=${EncodedChannel}${HOOK_ID}) [func_periodic_hook]
                    3. Set(GROUP(periodic-hook)=${GROUP_NAME})    [func_periodic_hook]
                    4. ExecIf($[${GROUP_COUNT(${GROUP_NAME}@periodic-hook)} > 1]?Hangup()) [func_periodic_hook]
                    5. Set(ChannelToSpy=${URIDECODE(${EncodedChannel})}) [func_periodic_hook]
                    6. ChanSpy(${ChannelToSpy},qEB)               [func_periodic_hook]

[ Context 'parkedcalls' created by 'res_parking/default' ]
  '700' =>          1. Park()                                     [res_parking]
  '701' =>          1. ParkedCall(default,701)                    [res_parking/default]
  '702' =>          1. ParkedCall(default,702)                    [res_parking/default]
  '703' =>          1. ParkedCall(default,703)                    [res_parking/default]
  '704' =>          1. ParkedCall(default,704)                    [res_parking/default]
  '705' =>          1. ParkedCall(default,705)                    [res_parking/default]
  '706' =>          1. ParkedCall(default,706)                    [res_parking/default]
  '707' =>          1. ParkedCall(default,707)                    [res_parking/default]
  '708' =>          1. ParkedCall(default,708)                    [res_parking/default]
  '709' =>          1. ParkedCall(default,709)                    [res_parking/default]
  '710' =>          1. ParkedCall(default,710)                    [res_parking/default]
  '711' =>          1. ParkedCall(default,711)                    [res_parking/default]
  '712' =>          1. ParkedCall(default,712)                    [res_parking/default]
  '713' =>          1. ParkedCall(default,713)                    [res_parking/default]
  '714' =>          1. ParkedCall(default,714)                    [res_parking/default]
  '715' =>          1. ParkedCall(default,715)                    [res_parking/default]
  '716' =>          1. ParkedCall(default,716)                    [res_parking/default]
  '717' =>          1. ParkedCall(default,717)                    [res_parking/default]
  '718' =>          1. ParkedCall(default,718)                    [res_parking/default]
  '719' =>          1. ParkedCall(default,719)                    [res_parking/default]
  '720' =>          1. ParkedCall(default,720)                    [res_parking/default]

[ Context 'ani' created by 'pbx_config' ]
  '_X.' =>          40000. NoOp(ANI: ${EXTEN})                    [pbx_config]
                    40001. Wait(0.25)                             [pbx_config]
                    40002. Answer()                               [pbx_config]
                    40003. Playback(vm-from)                      [pbx_config]
                    40004. SayDigits(${CALLERID(ani)})            [pbx_config]
                    40005. Wait(1.25)                             [pbx_config]
                    40006. SayDigits(${CALLERID(ani)})            [pbx_config]
                    40007. Return()                               [pbx_config]

[ Context 'time' created by 'pbx_config' ]
  '_X.' =>          30000. NoOp(Time: ${EXTEN} ${timezone})       [pbx_config]
                    30001. Wait(0.25)                             [pbx_config]
                    30002. Answer()                               [pbx_config]
                    30003. Set(FUTURETIME=$[${EPOCH} + 12])       [pbx_config]
                    30004. SayUnixTime(${FUTURETIME},Zulu,HNS)    [pbx_config]
                    30005. SayPhonetic(z)                         [pbx_config]
                    30006. SayUnixTime(${FUTURETIME},${timezone},HNS) [pbx_config]
                    30007. Playback(spy-local)                    [pbx_config]
                    30008. WaitUntil(${FUTURETIME})               [pbx_config]
                    30009. Playback(beep)                         [pbx_config]
                    30010. Return()                               [pbx_config]

[ Context 'default' created by 'pbx_config' ]
  Include =>        'demo'                                        [pbx_config]

[ Context 'public' created by 'pbx_config' ]
  Include =>        'demo'                                        [pbx_config]

[ Context 'page' created by 'pbx_config' ]
  '_X.' =>          1. Macro(page,SIP/${EXTEN})                   [pbx_config]

[ Context 'demo' created by 'pbx_config' ]
  '#' =>            1. Playback(demo-thanks)                      [pbx_config]
                    2. Hangup()                                   [pbx_config]
  '1000' =>         1. Goto(default,s,1)                          [pbx_config]
  '1234' =>         1. Playback(transfer,skip)                    [pbx_config]
                    2. Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)})) [pbx_config]
                    3. Goto(default,s,1)                          [pbx_config]
  '1235' =>         1. Voicemail(1234,u)                          [pbx_config]
  '1236' =>         1. Dial(Console/dsp)                          [pbx_config]
                    2. Voicemail(1234,b)                          [pbx_config]
  '2' =>            1. BackGround(demo-moreinfo)                  [pbx_config]
                    2. Goto(s,instruct)                           [pbx_config]
  '3' =>            1. Set(CHANNEL(language)=fr)                  [pbx_config]
                    2. Goto(s,restart)                            [pbx_config]
  '500' =>          1. Playback(demo-abouttotry)                  [pbx_config]
                    2. Dial(IAX2/guest@pbx.digium.com/s@default)  [pbx_config]
                    3. Playback(demo-nogo)                        [pbx_config]
                    4. Goto(s,6)                                  [pbx_config]
  '600' =>          1. Playback(demo-echotest)                    [pbx_config]
                    2. Echo()                                     [pbx_config]
                    3. Playback(demo-echodone)                    [pbx_config]
                    4. Goto(s,6)                                  [pbx_config]
  '76245' =>        1. Macro(page,SIP/Grandstream1)               [pbx_config]
  '7999' =>         1. Set(TIMEOUT(absolute)=60)                  [pbx_config]
                    2. Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d) [pbx_config]
  '8500' =>         1. VoicemailMain()                            [pbx_config]
                    2. Goto(s,6)                                  [pbx_config]
  'i' =>            1. Playback(invalid)                          [pbx_config]
  's' =>            1. Wait(1)                                    [pbx_config]
                    2. Answer()                                   [pbx_config]
                    3. Set(TIMEOUT(digit)=5)                      [pbx_config]
                    4. Set(TIMEOUT(response)=10)                  [pbx_config]
     [restart]      5. BackGround(demo-congrats)                  [pbx_config]
     [instruct]     6. BackGround(demo-instruct)                  [pbx_config]
                    7. WaitExten()                                [pbx_config]
  't' =>            1. Goto(#,1)                                  [pbx_config]
  '_7XXX' =>        1. Macro(page,SIP/${EXTEN})                   [pbx_config]
  Include =>        'stdexten'                                    [pbx_config]

[ Context 'macro-page' created by 'pbx_config' ]
  's' =>            1. ChanIsAvail(${ARG1},s)                     [pbx_config]
                    2. GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail) [pbx_config]
     [autoanswer]   3. Set(_ALERT_INFO="RA")                      [pbx_config]
                    4. SIPAddHeader(Call-Info: Answer-After=0)    [pbx_config]
                    5. NoOp()                                     [pbx_config]
                    6. Dial(${ARG1})                              [pbx_config]
     [fail]         7. Hangup()                                   [pbx_config]

[ Context 'stdPrivacyexten' created by 'pbx_config' ]
  'a' =>            1. VoicemailMain(${mbx})                      [pbx_config]
                    2. Return()                                   [pbx_config]
  'stdexten-BUSY' => 1. Voicemail(${mbx},b)                        [pbx_config]
                    2. NoOp(Finish stdPrivacyexten BUSY)          [pbx_config]
                    3. Return()                                   [pbx_config]
  'stdexten-DONTCALL' => 1. Goto(${dontcntx},s,1)                      [pbx_config]
  'stdexten-NOANSWER' => 1. Voicemail(${mbx},u)                        [pbx_config]
                    2. NoOp(Finish stdPrivacyexten NOANSWER)      [pbx_config]
                    3. Return()                                   [pbx_config]
  'stdexten-TORTURE' => 1. Goto(${tortcntx},s,1)                      [pbx_config]
  '_stde[x]te[n]-.' => 1. Goto(stdexten-NOANSWER,1)                  [pbx_config]
  '_X.' =>          60000. NoOp(Start stdPrivacyexten)            [pbx_config]
                    60001. Set(LOCAL(ext)=${ARG1})                [pbx_config]
                    60002. Set(LOCAL(dev)=${ARG2})                [pbx_config]
                    60003. Set(LOCAL(dontcntx)=${ARG3})           [pbx_config]
                    60004. Set(LOCAL(tortcntx)=${ARG4})           [pbx_config]
                    60005. Set(LOCAL(cntx)=${ARG5})               [pbx_config]
                    60006. Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""]) [pbx_config]
                    60007. Dial(${dev},20,p)                      [pbx_config]
                    60008. Goto(stdexten-${DIALSTATUS},1)         [pbx_config]

[ Context 'stdexten' created by 'pbx_config' ]
  'a' =>            1. VoicemailMain(${mbx})                      [pbx_config]
                    2. Return()                                   [pbx_config]
  'stdexten-BUSY' => 1. Voicemail(${mbx},b)                        [pbx_config]
                    2. Return()                                   [pbx_config]
  'stdexten-NOANSWER' => 1. Voicemail(${mbx},u)                        [pbx_config]
                    2. Return()                                   [pbx_config]
  '_stde[x]te[n]-.' => 1. Goto(stdexten-NOANSWER,1)                  [pbx_config]
  '_X.' =>          50000. NoOp(Start stdexten)                   [pbx_config]
                    50001. Set(LOCAL(ext)=${EXTEN})               [pbx_config]
                    50002. Set(LOCAL(dev)=${ARG1})                [pbx_config]
                    50003. Set(LOCAL(cntx)=${ARG2})               [pbx_config]
                    50004. Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})}) [pbx_config]
                    50005. Dial(${dev},20)                        [pbx_config]
                    50006. Goto(stdexten-${DIALSTATUS},1)         [pbx_config]

[ Context 'macro-trunkdial' created by 'pbx_config' ]
  's' =>            1. Dial(${ARG1})                              [pbx_config]
                    2. Goto(s-${DIALSTATUS},1)                    [pbx_config]
  's-BUSY' =>       1. Hangup()                                   [pbx_config]
  's-NOANSWER' =>   1. Hangup()                                   [pbx_config]
  '_s-.' =>         1. NoOp()                                     [pbx_config]

[ Context 'outbound-freenum2' created by 'pbx_config' ]
  'fn-BUSY' =>      1. Busy()                                     [pbx_config]
  '_f[n]-.' =>      1. NoOp(ISN: ${DIALSTATUS})                   [pbx_config]
                    2. Congestion()                               [pbx_config]
  '_X!' =>          1. Verbose(2,Performing ISN lookup for ${EXTEN}) [pbx_config]
                    2. Set(SUFFIX=${CUT(EXTEN,*,2-)})             [pbx_config]
                    3. GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1) [pbx_config]
                    4. Set(TIMEOUT(absolute)=10800)               [pbx_config]
                    5. Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) [pbx_config]
                    6. GotoIf($["${isnresult}" != ""]?from)       [pbx_config]
                    7. Set(DIALSTATUS=CONGESTION)                 [pbx_config]
                    8. Goto(fn-CONGESTION,1)                      [pbx_config]
     [from]         9. Set(__SIPFROMUSER=${CALLERID(num)})        [pbx_config]
                    10. GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) [pbx_config]
                    11. Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) [pbx_config]
     [dial]         12. Dial(SIP/${isnresult},40)                 [pbx_config]
                    13. Goto(fn-${DIALSTATUS},1)                  [pbx_config]

[ Context 'outbound-freenum' created by 'pbx_config' ]
  '_X*X!' =>        1. Goto(outbound-freenum2,${EXTEN},1)         [pbx_config]
  '_XX*X!' =>       1. Goto(outbound-freenum2,${EXTEN},1)         [pbx_config]
  '_XXX*X!' =>      1. Goto(outbound-freenum2,${EXTEN},1)         [pbx_config]
  '_XXXX*X!' =>     1. Goto(outbound-freenum2,${EXTEN},1)         [pbx_config]
  '_XXXXX*X!' =>    1. Goto(outbound-freenum2,${EXTEN},1)         [pbx_config]
  '_XXXXXX*X!' =>   1. Goto(outbound-freenum2,${EXTEN},1)         [pbx_config]

[ Context 'local' created by 'pbx_config' ]
  Include =>        'default'                                     [pbx_config]
  Include =>        'trunklocal'                                  [pbx_config]
  Include =>        'iaxtel700'                                   [pbx_config]
  Include =>        'trunktollfree'                               [pbx_config]
  Include =>        'iaxprovider'                                 [pbx_config]
  Include =>        'parkedcalls'                                 [pbx_config]
  Ignore pattern => '9'                                           [pbx_config]

[ Context 'longdistance' created by 'pbx_config' ]
  Include =>        'local'                                       [pbx_config]
  Include =>        'trunkld'                                     [pbx_config]
  Ignore pattern => '9'                                           [pbx_config]

[ Context 'international' created by 'pbx_config' ]
  Include =>        'longdistance'                                [pbx_config]
  Include =>        'trunkint'                                    [pbx_config]
  Ignore pattern => '9'                                           [pbx_config]

[ Context 'trunktollfree' created by 'pbx_config' ]
  '_91800NXXXXXX' => 1. Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [pbx_config]
  '_91866NXXXXXX' => 1. Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [pbx_config]
  '_91877NXXXXXX' => 1. Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [pbx_config]
  '_91888NXXXXXX' => 1. Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [pbx_config]

[ Context 'trunklocal' created by 'pbx_config' ]
  '_9NXXXXXX' =>    1. Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [pbx_config]

[ Context 'trunkld' created by 'pbx_config' ]
  '_91NXXNXXXXXX' => 1. Macro(dundi-e164,${EXTEN:1})               [pbx_config]
                    2. Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [pbx_config]

[ Context 'trunkint' created by 'pbx_config' ]
  '_9011.' =>       1. Macro(dundi-e164,${EXTEN:4})               [pbx_config]
                    2. Dial(${GLOBAL(TRUNK)}/${FILTER(0-9,${EXTEN:${GLOBAL(TRUNKMSD)}})}) [pbx_config]

[ Context 'iaxprovider' created by 'pbx_config' ]

[ Context 'iaxtel700' created by 'pbx_config' ]
  '_91700XXXXXXX' => 1. Dial(IAX2/${GLOBAL(IAXINFO)}@iaxtel.com/${EXTEN:1}@iaxtel) [pbx_config]

[ Context 'macro-dundi-e164' created by 'pbx_config' ]
  's' =>            1. Goto(${ARG1},1)                            [pbx_config]
  Include =>        'dundi-e164-lookup'                           [pbx_config]

[ Context 'dundi-e164-lookup' created by 'pbx_config' ]
  Include =>        'dundi-e164-local'                            [pbx_config]
  Include =>        'dundi-e164-switch'                           [pbx_config]

[ Context 'dundi-e164-switch' created by 'pbx_config' ]
  Alt. Switch =>    'DUNDi/e164'                                  [pbx_config]

[ Context 'dundi-e164-local' created by 'pbx_config' ]
  Include =>        'dundi-e164-canonical'                        [pbx_config]
  Include =>        'dundi-e164-customers'                        [pbx_config]
  Include =>        'dundi-e164-via-pstn'                         [pbx_config]

[ Context 'dundi-e164-via-pstn' created by 'pbx_config' ]

[ Context 'dundi-e164-customers' created by 'pbx_config' ]

[ Context 'dundi-e164-canonical' created by 'pbx_config' ]

-= 77 extensions (164 priorities) in 30 contexts. =-

vml000177*CLI> core set verbose 5

Console verbose was OFF and is now 5.

vml000177*CLI> 
    -- Accepting AUTHENTICATED call from 10.0.10.122:50548:
    --        > requested format = gsm,
    --        > requested prefs = (),
    --        > actual format = gsm,
    --        > host prefs = (gsm|ilbc|speex|g729|g723),
    --        > priority = mine
    -- Executing [600@default:1] Playback("IAX2/33-15362", "demo-echotest") in new stack
    -- <IAX2/33-15362> Playing 'demo-echotest.gsm' (language 'de')
    -- Executing [600@default:2] Echo("IAX2/33-15362", "") in new stack
    -- Executing [600@default:3] Playback("IAX2/33-15362", "demo-echodone") in new stack
    -- <IAX2/33-15362> Playing 'demo-echodone.gsm' (language 'de')
    -- Executing [600@default:4] Goto("IAX2/33-15362", "s,6") in new stack
    -- Goto (default,s,6)
    -- Executing [s@default:6] BackGround("IAX2/33-15362", "demo-instruct") in new stack
    -- <IAX2/33-15362> Playing 'demo-instruct.gsm' (language 'de')
    -- Executing [2@default:1] BackGround("IAX2/33-15362", "demo-moreinfo") in new stack
    -- <IAX2/33-15362> Playing 'demo-moreinfo.gsm' (language 'de')
  == Spawn extension (default, 2, 1) exited non-zero on 'IAX2/33-15362'
    -- Hungup 'IAX2/33-15362'

vml000177*CLI> 
    -- Accepting AUTHENTICATED call from 10.0.10.122:50548:
    --        > requested format = gsm,
    --        > requested prefs = (),
    --        > actual format = gsm,
    --        > host prefs = (gsm|ilbc|speex|g729|g723),
    --        > priority = mine
    -- Executing [500@default:1] Playback("IAX2/33-9804", "demo-abouttotry") in new stack
    -- <IAX2/33-9804> Playing 'demo-abouttotry.gsm' (language 'de')
    -- Executing [500@default:2] Dial("IAX2/33-9804", "IAX2/guest@pbx.digium.com/s@default") in new stack
    -- Called IAX2/guest@pbx.digium.com/s@default
[Jan 18 22:17:14] NOTICE[13560]: chan_iax2.c:4735 __auto_congest: Auto-congesting call due to slow response
    -- IAX2/216.207.245.8:4569-4356 is circuit-busy
    -- Hungup 'IAX2/216.207.245.8:4569-4356'
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [500@default:3] Playback("IAX2/33-9804", "demo-nogo") in new stack
    -- <IAX2/33-9804> Playing 'demo-nogo.gsm' (language 'de')
    -- Hungup 'IAX2/33-9804'

https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Dial

 # vim /etc/asterisk/extensions.conf

; interne Erreichbarkeit der Nebenstellen 11-99 untereinander herstellen
exten => _XX,1,NoOp(interner Teilnehmer mit der Rufnummer ${CALLERID(num)} ruft internen (PJ)SIP-Tln. ${EXTEN})
exten => _XX,n,Dial(PJSIP/${EXTEN},55,TtrWw)

 # asterisk -vvvvr

    -- Accepting AUTHENTICATED call from 88.217.123.90:33439:
    --        > requested format = gsm,
    --        > requested prefs = (),
    --        > actual format = gsm,
    --        > host prefs = (gsm|ilbc|speex|g729|g723),
    --        > priority = mine
    -- Executing [36@default:1] NoOp("IAX2/33-9632", "interner Teilnehmer mit der Rufnummer 33 ruft internen (PJ)SIP-Tln. 36") in new stack
    -- Executing [36@default:2] Dial("IAX2/33-9632", "PJSIP/36,55,TtrWw") in new stack
    -- Called PJSIP/36
    -- PJSIP/36-0000001b is ringing
  == Spawn extension (default, 36, 2) exited non-zero on 'IAX2/33-9632'
    -- Hungup 'IAX2/33-9632'

 # vim /etc/asterisk/extensions.conf

; Anwendungspeispiele bei ankommenden Amtsgesprächen

; BEISPIEL 1:
;            ankommende Anrufe vom Provider Telekom für Rufnummer 08121883175 auf der internen Nst. 36 signalisieren
;
exten => 08121883175,1,Verbose(### Anruf von ${CALLERID(num)} im Kontext "telekom_in" ###)
exten => 08121883175,n,Dial(PJSIP/36,60)
exten => 08121883175,n,Hangup

 # asterisk -vvvvr

    -- Executing [08121883175@telekom_in:1] Verbose("PJSIP/883172_in-0000001c", "### Anruf von +4917645534406 im Kontext "telekom_in" ###") in new stack
### Anruf von +4917645534406 im Kontext telekom_in ###
    -- Executing [08121883175@telekom_in:2] Dial("PJSIP/883172_in-0000001c", "PJSIP/36,60") in new stack
    -- Called PJSIP/36
    -- PJSIP/36-0000001d is ringing
  == Spawn extension (telekom_in, 08121883175, 2) exited non-zero on 'PJSIP/883172_in-0000001c'

 # vim /etc/asterisk/extensions.conf

; BEISPIEL 2:
;            ankommende Anrufe vom Provider Telekom für Rufnummer 08121883175 auf der internen Nst. 36 signalisieren und
;            bei nicht erreichen der Nebenstelle bzw. bei unbeantwortetem Anruf Weiterleitung zur Voicemail
;
exten => 08121883175,1,Verbose(### Anruf von ${CALLERID(num)} im Kontext "telekom_in" ###)
exten => 08121883175,n,Dial(PJSIP/36,60)
exten => 08121883175,n,Wait(1)
exten => 08121883175,n,VoiceMail(36,u)
exten => 08121883175,n,Hangup

 # asterisk -vvvvr

    -- Executing [08121883175@telekom_in:1] Verbose("PJSIP/883172_in-0000001e", "### Anruf von +4917645534406 im Kontext "telekom_in" ###") in new stack
### Anruf von +4917645534406 im Kontext telekom_in ###
    -- Executing [08121883175@telekom_in:2] Dial("PJSIP/883172_in-0000001e", "PJSIP/36,60") in new stack
    -- Called PJSIP/36
    -- PJSIP/36-0000001f is ringing
    -- Nobody picked up in 60000 ms
    -- Executing [08121883175@telekom_in:3] Wait("PJSIP/883172_in-0000001e", "1") in new stack
    -- Executing [08121883175@telekom_in:4] VoiceMail("PJSIP/883172_in-0000001e", "36,u") in new stack
    -- <PJSIP/883172_in-0000001e> Playing 'vm-theperson.alaw' (language 'de')
       > 0x7f647c01e910 -- Probation passed - setting RTP source address to 217.0.7.6:58666
    -- <PJSIP/883172_in-0000001e> Playing 'digits/3.alaw' (language 'de')
    -- <PJSIP/883172_in-0000001e> Playing 'digits/6.alaw' (language 'de')
    -- <PJSIP/883172_in-0000001e> Playing 'vm-isunavail.alaw' (language 'de')
    -- <PJSIP/883172_in-0000001e> Playing 'vm-intro.alaw' (language 'de')
    -- <PJSIP/883172_in-0000001e> Playing 'beep.alaw' (language 'de')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/36/tmp/QkCtOM format: wav, 0x7f648c00c0d8
    -- User ended message by pressing #
    -- <PJSIP/883172_in-0000001e> Playing 'auth-thankyou.alaw' (language 'de')
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
    -- Executing [08121883175@telekom_in:5] Hangup("PJSIP/883172_in-0000001e", "") in new stack
  == Spawn extension (telekom_in, 08121883175, 5) exited non-zero on 'PJSIP/883172_in-0000001e'

 # vim /etc/asterisk/extensions.conf

; -----------------------------------------------------------------------------------------------------------------------
; BEISPIEL 3:
;            ankommende Anrufe vom Provider Telekom für Rufnummer 08121883175 auf der internen Nst. 36 signalisieren und
;            bei nicht erreichen der Nebenstelle bzw. bei unbeantwortetem Anruf Weiterleitung zum Mobiltelefon
;
exten => 08121883175,1,Verbose(### Anruf von ${CALLERID(num)} im Kontext "telekom_in" ###)
exten => 08121883175,n,Dial(PJSIP/36,10)
exten => 08121883175,n,Wait(1)
exten => 08121883175,n,Dial(PJSIP/883176_out/sip:017645534406@tel.t-online.de,20,rWw)
exten => 08121883175,n,Hangup
; ---------------------------------------------------------------------------------------------------------------------

 # asterisk -vvvvr

    -- Executing [08121883175@telekom_in:1] Verbose("PJSIP/883172_in-00000000", "### Anruf von anonymous im Kontext "telekom_in" ###") in new stack
### Anruf von anonymous im Kontext telekom_in ###
    -- Executing [08121883175@telekom_in:2] Dial("PJSIP/883172_in-00000000", "PJSIP/36,10") in new stack
    -- Called PJSIP/36
  == Spawn extension (telekom_in, 08121883175, 2) exited non-zero on 'PJSIP/883172_in-00000000'
    -- Executing [08121883175@telekom_in:1] Verbose("PJSIP/883172_in-00000002", "### Anruf von anonymous im Kontext "telekom_in" ###") in new stack
### Anruf von anonymous im Kontext telekom_in ###
    -- Executing [08121883175@telekom_in:2] Dial("PJSIP/883172_in-00000002", "PJSIP/36,10") in new stack
    -- Called PJSIP/36
    -- PJSIP/36-00000003 is ringing
    -- Nobody picked up in 10000 ms
    -- Executing [08121883175@telekom_in:3] Wait("PJSIP/883172_in-00000002", "1") in new stack
    -- Executing [08121883175@telekom_in:4] Dial("PJSIP/883172_in-00000002", "PJSIP/883176_out/sip:017645534406@tel.t-online.de,20,rWw") in new stack
    -- Called PJSIP/883176_out/sip:017645534406@tel.t-online.de
    -- PJSIP/883176_out-00000004 is making progress passing it to PJSIP/883172_in-00000002
    -- PJSIP/883176_out-00000004 is ringing
    -- PJSIP/883176_out-00000004 answered PJSIP/883172_in-00000002
    -- Channel PJSIP/883176_out-00000004 joined 'simple_bridge' basic-bridge <19dd0b6d-7cc4-4e8b-850a-e932484bfd20>
    -- Channel PJSIP/883172_in-00000002 joined 'simple_bridge' basic-bridge <19dd0b6d-7cc4-4e8b-850a-e932484bfd20>
       > Bridge 19dd0b6d-7cc4-4e8b-850a-e932484bfd20: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'PJSIP/883172_in-00000002' and 'PJSIP/883176_out-00000004' - media will flow directly between them
       > Remotely bridged 'PJSIP/883172_in-00000002' and 'PJSIP/883176_out-00000004' - media will flow directly between them
    -- Channel PJSIP/883176_out-00000004 left 'native_rtp' basic-bridge <19dd0b6d-7cc4-4e8b-850a-e932484bfd20>
    -- Channel PJSIP/883172_in-00000002 left 'native_rtp' basic-bridge <19dd0b6d-7cc4-4e8b-850a-e932484bfd20>
  == Spawn extension (telekom_in, 08121883175, 4) exited non-zero on 'PJSIP/883172_in-00000002'

 # vim /etc/asterisk/extensions.conf

; -----------------------------------------------------------------------------------------------------------------------
; BEISPIEL 4:
;            ankommende Anrufe vom Provider Telekom für Rufnummer 08121883175 auf der internen Nst. 36 und zeitgleich
;            auf dem Mobiltelefon signalisieren. Anschließend bei nicht erreichen der Nebenstelle bzw. bei 
;            unbeantwortetem Anruf Weiterleitung zur Voicemailbox. Leistungsmerkmal "twinning"
;
exten => 08121883175,1,Verbose(### Anruf von ${CALLERID(num)} im Kontext "telekom_in" ###)
exten => 08121883175,n,Dial(PJSIP/36&PJSIP/883176_out/sip:017645534406@tel.t-online.de,10,rWw)
exten => 08121883175,n,Wait(1)
exten => 08121883175,n,VoiceMail(36,u)
exten => 08121883175,n,Hangup
; -----------------------------------------------------------------------------------------------------------------------

 # asterisk -vvvvr

    -- Executing [08121883175@telekom_in:1] Verbose("PJSIP/883172_in-00000000", "### Anruf von anonymous im Kontext "telekom_in" ###") in new stack
### Anruf von anonymous im Kontext telekom_in ###
    -- Executing [08121883175@telekom_in:2] Dial("PJSIP/883172_in-00000000", "PJSIP/36&PJSIP/883176_out/sip:017645534406@tel.t-online.de,10,rWw") in new stack
    -- Called PJSIP/36
    -- Called PJSIP/883176_out/sip:017645534406@tel.t-online.de
    -- PJSIP/36-00000001 connected line has changed. Saving it until answer for PJSIP/883172_in-00000000
    -- PJSIP/883176_out-00000002 connected line has changed. Saving it until answer for PJSIP/883172_in-00000000
    -- PJSIP/36-00000001 is ringing
    -- PJSIP/883176_out-00000002 is making progress passing it to PJSIP/883172_in-00000000
    -- PJSIP/883176_out-00000002 is ringing
    -- Nobody picked up in 10000 ms
    -- Executing [08121883175@telekom_in:3] Wait("PJSIP/883172_in-00000000", "1") in new stack
    -- Executing [08121883175@telekom_in:4] VoiceMail("PJSIP/883172_in-00000000", "36,u") in new stack
    -- <PJSIP/883172_in-00000000> Playing 'vm-theperson.alaw' (language 'de')
       > 0x7f47f0023ce0 -- Probation passed - setting RTP source address to 217.0.4.196:12838
    -- <PJSIP/883172_in-00000000> Playing 'digits/3.alaw' (language 'de')
    -- <PJSIP/883172_in-00000000> Playing 'digits/6.alaw' (language 'de')
    -- <PJSIP/883172_in-00000000> Playing 'vm-isunavail.alaw' (language 'de')
    -- <PJSIP/883172_in-00000000> Playing 'vm-intro.alaw' (language 'de')
    -- <PJSIP/883172_in-00000000> Playing 'beep.alaw' (language 'de')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/36/tmp/ObhJK5 format: wav, 0x7f47f4008488
    -- User hung up
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Spawn extension (telekom_in, 08121883175, 4) exited non-zero on 'PJSIP/883172_in-00000000'

 # vim /etc/asterisk/extensions.conf

; -----------------------------------------------------------------------------------------------------------------------
; BEISPIEL 5:
;            ankommende Anrufe vom Provider Telekom für Rufnummer 08121883175 nur dann auf der internen Nst. 36
;            signalisieren, wenn der Anruf _NICHT_ vom Anschluss +49892165... kommt. Anrufer dieses dieser Anschlusses
;            werden zur Abwimmelansage "... dieser Anschluss ist vorübergehend nicht erreichbar ..." weitergeleitet.
;            Alle anderen Anrufer werden zur Nst. 36 durchgestellt; bzw. an ie Voicemail weitergeleitet
exten => 08121883175/_+49892165.,1,NoOp(Externanrufer-Gruppe ${CALLERID(num)} ist geblacklistet und wird daher auf die Voicemail geleitet!)
exten => 08121883175/_+49892165.,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175/_+49892165.,n,wait(1)
exten => 08121883175/_+49892165.,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175/_+49892165.,n,wait(1)
exten => 08121883175/_+49892165.,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175/_+49892165.,n,wait(1)
exten => 08121883175/_+49892165.,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175/_+49892165.,n,wait(1)
exten => 08121883175/_+49892165.,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175/_+49892165.,n,Hangup
exten => 08121883175,1,NoOp(Externanrufer ${CALLERID(num)} hat _keinen_ Sperreintrag und wird an den Nst. 36 (snom 360))
exten => 08121883175,n,Dial(PJSIP/36,20,rWw)
exten => 08121883175,n,Wait(1)  
exten => 08121883175,n,VoiceMail(36,u)
exten => 08121883175,n,Hangup
; -----------------------------------------------------------------------------------------------------------------------

 # asterisk -vvvvr

    -- Executing [08121883175@telekom_in:1] NoOp("PJSIP/883172_in-00000000", "Externanrufer-Gruppe +49892162407 ist geblacklistet und wird daher auf die Voicemail geleitet!") in new stack
    -- Executing [08121883175@telekom_in:2] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
       > 0x7f9e7803f350 -- Probation passed - setting RTP source address to 217.0.4.230:13512
    -- Executing [08121883175@telekom_in:3] Wait("PJSIP/883172_in-00000000", "1") in new stack
    -- Executing [08121883175@telekom_in:4] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
    -- Executing [08121883175@telekom_in:5] Wait("PJSIP/883172_in-00000000", "1") in new stack
    -- Executing [08121883175@telekom_in:6] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
    -- Executing [08121883175@telekom_in:7] Wait("PJSIP/883172_in-00000000", "1") in new stack
    -- Executing [08121883175@telekom_in:8] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
    -- Executing [08121883175@telekom_in:9] Wait("PJSIP/883172_in-00000000", "1") in new stack
    -- Executing [08121883175@telekom_in:10] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
    -- Executing [08121883175@telekom_in:11] Hangup("PJSIP/883172_in-00000000", "") in new stack
  == Spawn extension (telekom_in, 08121883175, 11) exited non-zero on 'PJSIP/883172_in-00000000'

    -- Executing [08121883175@telekom_in:1] NoOp("PJSIP/883172_in-00000001", "Externanrufer anonymous hat _keinen_ Sperreintrag und wird an den Nst. 36 (snom 360)") in new stack
    -- Executing [08121883175@telekom_in:2] Dial("PJSIP/883172_in-00000001", "PJSIP/36,20,rWw") in new stack
    -- Called PJSIP/36
    -- PJSIP/36-00000002 is ringing
    -- Nobody picked up in 20000 ms
    -- Executing [08121883175@telekom_in:3] Wait("PJSIP/883172_in-00000001", "1") in new stack
    -- Executing [08121883175@telekom_in:4] VoiceMail("PJSIP/883172_in-00000001", "36,u") in new stack
    -- <PJSIP/883172_in-00000001> Playing 'vm-theperson.alaw' (language 'de')
       > 0x7f9e5800abb0 -- Probation passed - setting RTP source address to 217.0.7.70:28670
    -- <PJSIP/883172_in-00000001> Playing 'digits/3.alaw' (language 'de')
    -- <PJSIP/883172_in-00000001> Playing 'digits/6.alaw' (language 'de')
    -- <PJSIP/883172_in-00000001> Playing 'vm-isunavail.alaw' (language 'de')
    -- <PJSIP/883172_in-00000001> Playing 'vm-intro.alaw' (language 'de')
    -- <PJSIP/883172_in-00000001> Playing 'beep.alaw' (language 'de')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/36/tmp/7SpfZs format: wav, 0x7f9e6c3f6508
    -- User hung up
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Spawn extension (telekom_in, 08121883175, 4) exited non-zero on 'PJSIP/883172_in-00000001'

 # vim /etc/asterisk/extensions.conf

; -----------------------------------------------------------------------------------------------------------------------
; BEISPIEL 6:
;            ankommende Anrufe vom Provider Telekom für Rufnummer 08121883175 nur dann auf der internen Nst. 36
;            signalisieren, wenn der Anruf _NICHT_ die Rufnummer +498921652407 überträgt. Dieser Anrufer wird sofort
;            an die Voicemail weitergeleitet. Alle anderen Anrufer werden zur Nst. 36 durchgestellt und bei nicht 
;            erreichen der Nebenstelle bzw. bei unbeantwortetem Anruf rfolgt eine Weiterleitung zur Voicemailbox.
;
exten => 08121883175,1,Verbose(### Anruf von ${CALLERID(num)} im Kontext "telekom_in" ###)
exten => 08121883175,n,GotoIf($["${CALLERID(num)}" = "+498921652407"]?10:20)
exten => 08121883175,10,NoOp(Externanrufer ${CALLERID(num)} ist geblacklistet und wird daher auf die Voicemail geleitet!)
exten => 08121883175,n,VoiceMail(36,u)
exten => 08121883175,n,Hangup
exten => 08121883175,20,NoOp(Externanrufer ${CALLERID(num)} hat keinen Sperreintrag und wird an Nst. 36 signalisiert!)
exten => 08121883175,n,Dial(PJSIP/36,20,rWw)
exten => 08121883175,n,Wait(1)
exten => 08121883175,n,VoiceMail(36,u)
exten => 08121883175,n,Hangup
; -----------------------------------------------------------------------------------------------------------------------

 # asterisk -vvvvr

    -- Executing [08121883175@telekom_in:1] Verbose("PJSIP/883172_in-00000000", "### Anruf von +498921652407 im Kontext "telekom_in" ###") in new stack
### Anruf von +498921652407 im Kontext telekom_in ###
    -- Executing [08121883175@telekom_in:2] GotoIf("PJSIP/883172_in-00000000", "1?10:20") in new stack
    -- Goto (telekom_in,08121883175,10)
    -- Executing [08121883175@telekom_in:10] NoOp("PJSIP/883172_in-00000000", "Externanrufer +498921652407 ist geblacklistet und wird daher auf die Voicemail geleitet!") in new stack
    -- Executing [08121883175@telekom_in:11] VoiceMail("PJSIP/883172_in-00000000", "36,u") in new stack
    -- <PJSIP/883172_in-00000000> Playing 'vm-theperson.alaw' (language 'de')
       > 0x7fd2d0042a80 -- Probation passed - setting RTP source address to 217.0.7.4:10530
    -- <PJSIP/883172_in-00000000> Playing 'digits/3.alaw' (language 'de')
    -- <PJSIP/883172_in-00000000> Playing 'digits/6.alaw' (language 'de')
    -- <PJSIP/883172_in-00000000> Playing 'vm-isunavail.alaw' (language 'de')
    -- <PJSIP/883172_in-00000000> Playing 'vm-intro.alaw' (language 'de')
    -- <PJSIP/883172_in-00000000> Playing 'beep.alaw' (language 'de')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/36/tmp/jAEk5n format: wav, 0x7fd2e0008058
    -- User hung up
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Parsing '/var/spool/asterisk/voicemail/default/36/INBOX/msg0000.txt': Found
  == Spawn extension (telekom_in, 08121883175, 11) exited non-zero on 'PJSIP/883172_in-00000000'

    -- Executing [08121883175@telekom_in:1] Verbose("PJSIP/883172_in-00000001", "### Anruf von anonymous im Kontext "telekom_in" ###") in new stack
### Anruf von anonymous im Kontext telekom_in ###
    -- Executing [08121883175@telekom_in:2] GotoIf("PJSIP/883172_in-00000001", "0?10:20") in new stack
    -- Goto (telekom_in,08121883175,20)
    -- Executing [08121883175@telekom_in:20] NoOp("PJSIP/883172_in-00000001", "Externanrufer anonymous hat keinen Sperreintrag und wird an Nst. 36 signalisiert!") in new stack
    -- Executing [08121883175@telekom_in:21] Dial("PJSIP/883172_in-00000001", "PJSIP/36,20,rWw") in new stack
    -- Called PJSIP/36
    -- PJSIP/36-00000002 is ringing
  == Spawn extension (telekom_in, 08121883175, 21) exited non-zero on 'PJSIP/883172_in-00000001'

 # vim /etc/asterisk/extensions.conf

; -----------------------------------------------------------------------------------------------------------------------
; BEISPIEL 7:
;            alle ankommende Anrufe vom Provider Telekom für Rufnummer 08121883175 werden in der 1st-level Supportgruppe,
;            d.h. an den Nebenstellen 33 und 36 signalisiert. Ruf hingegen unser VIP-Kunde mit der Rufnummer 08931408448
;            wird dieser Anrufer sofort zum 2nd-level Support (Nst. 31) durchgestellt bzw. bei nicht erreichen zum 
;            Mobiltelefon durchgestellt.  
;            
exten => 08121883175/_+498931408448,1,Verbose(### Anruf von VIP-Kunden mit der Rufnummer ${CALLERID(num)} im Kontext "telekom_in". VIP-Behandlung veranlassen! ###)
exten => 08121883175/_+498931408448,n,Dial(PJSIP/31&PJSIP/883176_out/sip:017645534406@tel.t-online.de,10,rWw)
exten => 08121883175/_+498931408448,n,Hangup
exten => 08121883175,1,NoOp(Externanrufer mit der Rufnummer ${CALLERID(num)} ist 08/15-Kunde und wird im 1st-level Support abgeworfen.)
exten => 08121883175,n,Dial(PJSIP/36&IAX2/33,15)
exten => 08121883175,n,Wait(1)
exten => 08121883175,n,VoiceMail(36,u)
exten => 08121883175,n,Hangup
; -----------------------------------------------------------------------------------------------------------------------

    -- Executing [08121883175@telekom_in:1] Verbose("PJSIP/883172_in-00000003", "### Anruf von VIP-Kunden mit der Rufnummer +4989216750916 im Kontext "telekom_in". VIP-Behandlung veranlassen! ###") in new stack
### Anruf von VIP-Kunden mit der Rufnummer +4989216750916 im Kontext telekom_in. VIP-Behandlung veranlassen! ###
    -- Executing [08121883175@telekom_in:2] Dial("PJSIP/883172_in-00000003", "PJSIP/31&PJSIP/883176_out/sip:017645534406@tel.t-online.de,10,rWw") in new stack
    -- Called PJSIP/31
    -- Called PJSIP/883176_out/sip:017645534406@tel.t-online.de
    -- PJSIP/31-00000004 connected line has changed. Saving it until answer for PJSIP/883172_in-00000003
    -- PJSIP/883176_out-00000005 connected line has changed. Saving it until answer for PJSIP/883172_in-00000003
    -- PJSIP/31-00000004 is ringing
    -- PJSIP/883176_out-00000005 is making progress passing it to PJSIP/883172_in-00000003
    -- PJSIP/883176_out-00000005 is ringing
  == Spawn extension (telekom_in, 08121883175, 2) exited non-zero on 'PJSIP/883172_in-00000003'

    -- Executing [08121883175@telekom_in:1] NoOp("PJSIP/883172_in-00000000", "Externanrufer mit der Rufnummer anonymous ist 08/15-Kunde und wird im 1st-level Support abgeworfen.") in new stack
    -- Executing [08121883175@telekom_in:2] Dial("PJSIP/883172_in-00000000", "PJSIP/36&IAX2/33,15") in new stack
[Jan 30 15:31:12] WARNING[17010][C-00000001]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Subscriber absent)
    -- Called PJSIP/36
    -- PJSIP/36-00000001 is ringing
  == Spawn extension (telekom_in, 08121883175, 2) exited non-zero on 'PJSIP/883172_in-00000000'

 # vim /etc/asterisk/extensions.conf

; -----------------------------------------------------------------------------------------------------------------------
; BEISPIEL 8:
;            alle ankommende Anrufe vom Provider Telekom für Rufnummer 08121883175 werden gegen die black- und white-
;            listing Datenbank geprüft und gesondert behandelt: 
;              - Anrufer die keine Nummer übertragen werden "nur" an der int. Nst 36 signalisiert, ohne Weiterleitung 
;                zum Händie und/oder Voicemail
;              - Anrufer mit einem Whitelisting-Eintrag in der AstDB (Asterisk-Datenbank) werden sowohl an der internen 
;                Nebenstelle 36 wie auch am Händie mit der Rufnummer 0176/45534406 signalisiert. Wird der Anrufer nicht 
;                binnen 20 Sekunden beantwortet, erhält der Anrufer die Möglichkeit zum Hinterlassen einer Nachricht 
;                auf der Voicemailbox von der Nst. 36
;              - Anrufer, die einen blacklisting Eintrag haben, werden direkt zur Abwimmelansage der Nst. 36 geleitet 
;                und anschließend die Verbindung gekappt, ohne dass der Anrufer eine Nachricht hinterlassen kann.
;
exten => 08121883175,1,NoOp(Anrufer übertrug die Rufnummer ${CALLERID(num)})
exten => 08121883175,n,Set(access=${DB(36/${CALLERID(num)})})

exten => 08121883175,n,GotoIf($[ "${access}" = "" ]?ohne:testing)
exten => 08121883175,n(testing),GotoIf($[ "${access}" = "wl" ]?whitelisted:blacklisted)
exten => 08121883175,n(whitelisted),NoOp(der Anrufer ist gewhitelisted, der Anrufer wird zur Nebenstelle 36 und zum Mobiltelefon durchgestellt)
exten => 08121883175,n,Dial(PJSIP/36&PJSIP/883176_out/sip:017645534406@tel.t-online.de,10,rWw)
exten => 08121883175,n,Wait(1)
exten => 08121883175,n,VoiceMail(36,u)
exten => 08121883175,n,Hangup
exten => 08121883175,n(blacklisted),NoOp(der Anrufer ist geblacklisted und wird daher zur "Abwimmel"-Voicemailbox abgeworfen)
exten => 08121883175,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175,n,wait(1)
exten => 08121883175,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175,n,wait(1)
exten => 08121883175,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175,n,wait(1)
exten => 08121883175,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175,n,wait(1)
exten => 08121883175,n,Playback(/var/spool/asterisk/voicemail/default/31/unerreichbar)
exten => 08121883175,n,Hangup
exten => 08121883175,n(ohne),NoOp(der Anrufer überträgt keine Nummer, und ist weder geblack- noch gewhitelisted)
exten => 08121883175,n,Dial(PJSIP/31,60)
exten => 08121883175,n,Hangup
; -----------------------------------------------------------------------------------------------------------------------

 # asterisk -vvvvr

    -- Executing [08121883175@telekom_in:1] NoOp("PJSIP/883172_in-0000000b", "Anrufer übertrug die Rufnummer +4917645534406") in new stack
    -- Executing [08121883175@telekom_in:2] Set("PJSIP/883172_in-0000000b", "access=wl") in new stack
    -- Executing [08121883175@telekom_in:3] GotoIf("PJSIP/883172_in-0000000b", "0?ohne:testing") in new stack
    -- Goto (telekom_in,08121883175,4)
    -- Executing [08121883175@telekom_in:4] GotoIf("PJSIP/883172_in-0000000b", "1?whitelisted:blacklisted") in new stack
    -- Goto (telekom_in,08121883175,5)
    -- Executing [08121883175@telekom_in:5] NoOp("PJSIP/883172_in-0000000b", "der Anrufer ist gewhitelisted, der Anrufer wird zur Nebenstelle 36 und zum Mobiltelefon durchgestellt") in new stack
    -- Executing [08121883175@telekom_in:6] Dial("PJSIP/883172_in-0000000b", "PJSIP/36&PJSIP/883176_out/sip:017645534406@tel.t-online.de,10,rWw") in new stack
    -- Called PJSIP/36
    -- Called PJSIP/883176_out/sip:017645534406@tel.t-online.de
    -- PJSIP/36-0000000c connected line has changed. Saving it until answer for PJSIP/883172_in-0000000b
    -- PJSIP/883176_out-0000000d connected line has changed. Saving it until answer for PJSIP/883172_in-0000000b
    -- PJSIP/36-0000000c is ringing
    -- PJSIP/883176_out-0000000d is making progress passing it to PJSIP/883172_in-0000000b
    -- PJSIP/883176_out-0000000d is ringing
    -- PJSIP/883176_out-0000000d answered PJSIP/883172_in-0000000b
    -- Channel PJSIP/883176_out-0000000d joined 'simple_bridge' basic-bridge <868dad5d-a999-4b8e-babf-88fe30fbf9af>
    -- Channel PJSIP/883172_in-0000000b joined 'simple_bridge' basic-bridge <868dad5d-a999-4b8e-babf-88fe30fbf9af>
       > Bridge 868dad5d-a999-4b8e-babf-88fe30fbf9af: switching from simple_bridge technology to native_rtp
       > Remotely bridged 'PJSIP/883172_in-0000000b' and 'PJSIP/883176_out-0000000d' - media will flow directly between them
       > Remotely bridged 'PJSIP/883172_in-0000000b' and 'PJSIP/883176_out-0000000d' - media will flow directly between them
    -- Channel PJSIP/883172_in-0000000b left 'native_rtp' basic-bridge <868dad5d-a999-4b8e-babf-88fe30fbf9af>
  == Spawn extension (telekom_in, 08121883175, 6) exited non-zero on 'PJSIP/883172_in-0000000b'
    -- Channel PJSIP/883176_out-0000000d left 'native_rtp' basic-bridge <868dad5d-a999-4b8e-babf-88fe30fbf9af>

 # asterisk -vvvvr

    -- Executing [08121883175@telekom_in:1] NoOp("PJSIP/883172_in-00000000", "Anrufer übertrug die Rufnummer +4917645502006") in new stack
    -- Executing [08121883175@telekom_in:2] Set("PJSIP/883172_in-00000000", "access=bl") in new stack
    -- Executing [08121883175@telekom_in:3] GotoIf("PJSIP/883172_in-00000000", "0?ohne:testing") in new stack
    -- Goto (telekom_in,08121883175,4)
    -- Executing [08121883175@telekom_in:4] GotoIf("PJSIP/883172_in-00000000", "0?whitelisted:blacklisted") in new stack
    -- Goto (telekom_in,08121883175,10)
    -- Executing [08121883175@telekom_in:10] NoOp("PJSIP/883172_in-00000000", "der Anrufer ist geblacklisted und wird daher zur "Abwimmel"-Voicemailbox abgeworfen") in new stack
    -- Executing [08121883175@telekom_in:11] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
       > 0x7f2f3402ac30 -- Probation passed - setting RTP source address to 217.0.4.68:48282
    -- Executing [08121883175@telekom_in:12] Wait("PJSIP/883172_in-00000000", "1") in new stack
    -- Executing [08121883175@telekom_in:13] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
    -- Executing [08121883175@telekom_in:14] Wait("PJSIP/883172_in-00000000", "1") in new stack
    -- Executing [08121883175@telekom_in:15] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
    -- Executing [08121883175@telekom_in:16] Wait("PJSIP/883172_in-00000000", "1") in new stack
    -- Executing [08121883175@telekom_in:17] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
    -- Executing [08121883175@telekom_in:18] Wait("PJSIP/883172_in-00000000", "1") in new stack
    -- Executing [08121883175@telekom_in:19] Playback("PJSIP/883172_in-00000000", "/var/spool/asterisk/voicemail/default/31/unerreichbar") in new stack
    -- <PJSIP/883172_in-00000000> Playing '/var/spool/asterisk/voicemail/default/31/unerreichbar.slin' (language 'de')
    -- Executing [08121883175@telekom_in:20] Hangup("PJSIP/883172_in-00000000", "") in new stack
  == Spawn extension (telekom_in, 08121883175, 20) exited non-zero on 'PJSIP/883172_in-00000000'

 # asterisk -vvvvr

    -- Executing [08121883175@telekom_in:1] NoOp("PJSIP/883172_in-00000009", "Anrufer übertrug die Rufnummer anonymous") in new stack
    -- Executing [08121883175@telekom_in:2] Set("PJSIP/883172_in-00000009", "access=") in new stack
    -- Executing [08121883175@telekom_in:3] GotoIf("PJSIP/883172_in-00000009", "1?ohne:testing") in new stack
    -- Goto (telekom_in,08121883175,21)
    -- Executing [08121883175@telekom_in:21] NoOp("PJSIP/883172_in-00000009", "der Anrufer überträgt keine Nummer, und ist weder geblack- noch gewhitelisted") in new stack
    -- Executing [08121883175@telekom_in:22] Dial("PJSIP/883172_in-00000009", "PJSIP/31,60") in new stack
    -- Called PJSIP/31
    -- PJSIP/31-0000000a is ringing
  == Spawn extension (telekom_in, 08121883175, 22) exited non-zero on 'PJSIP/883172_in-00000009'

https://www.asterisksounds.org/de/download

 # mkdir /usr/share/asterisk/sounds/de

 # de /usr/local/src

 # wget https://www.asterisksounds.org/de/download/asterisk-sounds-core-de-sln16.zip
 # wget https://www.asterisksounds.org/de/download/asterisk-sounds-extra-de-sln16.zip

 # unzip /usr/local/src/asterisk-sounds-core-de-sln16.zip -d /usr/share/asterisk/sounds/de/
 # unzip /usr/local/src/asterisk-sounds-extra-de-sln16.zip -d /usr/share/asterisk/sounds/de/

 # chmod 644 /usr/share/asterisk/sounds/de/* -R

 # find /usr/share/asterisk/sounds/de -type d -exec chmod 0755 {} \;

 # yum install sox

 # vi /usr/local/bin/asterisk-sound-convert

/usr/local/bin/asterisk-sound-convert

#!/bin/bash
cd /usr/share/asterisk/sounds/de
for a in $(find . -name '*.sln16'); do
  sox -t raw -e signed-integer -b 16 -c 1 -r 16k $a -t gsm -r 8k `echo $a|sed "s/.sln16/.gsm/"`;\
  sox -t raw -e signed-integer -b 16 -c 1 -r 16k $a -t raw -r 8k -e a-law `echo $a|sed "s/.sln16/.alaw/"`;\
  sox -t raw -e signed-integer -b 16 -c 1 -r 16k $a -t raw -r 8k -e mu-law `echo $a|sed "s/.sln16/.ulaw/"`;\
done

 # chmod +x /usr/local/bin/asterisk-sound-convert

 # asterisk-sound-convert

 # yum install madplay -y

https://www.asteriskguru.com/tools/audio_conversion.php

Haben wir von unserem VoIP-Anbieter mehrere Rufnummern bekommen, so werden wir diese in aller regel einzelnen Endgeräten oder Diensten zuordnen wollen. Bei ankommenden Gesprächen haben wir diese Zuordnung über den Abschnitt [telekom_in] vorgenommen. Nun wollen wir aber auch je nach verwendeter internen Nebenstelle unseres Asterisk eine bestimmte Rufnummr CLID¹⁾ als Absender nach extern verwenden und auch übertragen. Das was sich bei dem ein oder anderen höchstprofesssionellen Systemintegrator als unlösbare Herausforderunge bei einer VoIP-Anlage mit 3.000 Nebenstellen darstellen kann, gestaltet sich bei unserem Asterisk-Server als triviale Herausforderung.

Für die Zuordnung der internen Rufnummern zu den jeweiligen externen Absenderrufnummern bzw. PJSIP-Kanälen bedienen wir uns der Asterisk-eigenen Datenbank. Diese Datenbank erreichen wir über die Funktion database. Wir können damit z.B. unsere Registrierten Geräte abfragen.

 # asterisk -r

 vml000177*CLI> database put 33 CLID 9959277_out

Updated database successfully

 vml000177*CLI> database put 36 CLID 9959278_out

Updated database successfully

 vml000177*CLI> database show

/33/CLID                                          : 9959277_out              
/36/CLID                                          : 9959278_out              

 # yum install asterisk-voicemail -y

 # chown asterisk:asterisk /var/spool/asterisk/voicemail/ -R

 # wget https://forensics.cert.org/cert-forensics-tools-release-el7.rpm
 # rpm -Uvh cert-forensics-tools-release*rpm

 # vim /etc/yum.repos.d/cert-forensics-tools.repo

/etc/yum.repos.d/cert-forensics-tools.repo

[forensics]
name=CERT Forensics Tools Repository
baseurl=http://www.cert.org/forensics/repository/centos/cert/$releasever/$basearch
enabled=1
gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-cert-forensics-2018-04-07
gpgcheck=1
priority=10
 
[forensics-test]
name=CERT Forensics Tools Test Repository
baseurl=http://www.cert.org/forensics/repository/centos/cert-test/$releasever/$basearch
enabled=0
gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-cert-forensics-2018-04-07
gpgcheck=1
 
[forensics-splunk]
name=CERT Forensics Tools Repository - Splunk
baseurl=http://www.cert.org/forensics/repository/centos/splunk/$releasever/$basearch/
gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-splunk
enabled=0
gpgcheck=1
 
[forensics-sip]
name=CERT Forensics Tools Repository - SiLK, IPA, and Postgresql
baseurl=http://www.cert.org/forensics/repository/centos/sip/$releasever/$basearch
enabled=0
gpgkey=file:///etc/pki/rpm-gpg/RPM-GPG-KEY-cert-forensics-2018-04-07
gpgcheck=1

 # yum install lame -y

 # yum install asterisk-fax

 # rpm -qil asterisk-fax

Name        : asterisk-fax
Version     : 14.2.1
Release     : 1.el7.centos
Architecture: x86_64
Install Date: Sun 29 Jan 2017 09:51:54 PM CET
Group       : Applications/Internet
Size        : 155968
License     : GPLv2
Signature   : RSA/SHA1, Thu 08 Dec 2016 11:58:07 PM CET, Key ID ccf40e16b9a46fa9
Source RPM  : asterisk-14.2.1-1.el7.centos.src.rpm
Build Date  : Thu 08 Dec 2016 11:57:54 PM CET
Build Host  : olive.ph.tucny.com
Relocations : (not relocatable)
URL         : http://www.asterisk.org/
Summary     : FAX applications for Asterisk
Description :
FAX applications for Asterisk
/etc/asterisk/res_fax.conf
/usr/lib64/asterisk/modules/res_fax.so
/usr/lib64/asterisk/modules/res_fax_spandsp.so