Unterschiede

Hier werden die Unterschiede zwischen zwei Versionen angezeigt.

--- voip:c7:asterisk14:start [18.01.2017 14:23. ] – [ast.tucny.com] django
+++ voip:c7:asterisk14:start [18.11.2024 06:56. ] (aktuell) – Externe Bearbeitung 127.0.0.1
@@ Zeile 2: / Zeile 2: @@
 {{voip:asterisk.gif|Asterisk Logo}}
 Asterisk ist eine OpenSource Telekommunkationssystem, entwickelt von Marc Spencer / Digium für Linux Systeme.
+===== Dokumentation =====
+Hinweise zum Release **14** finden sich auf der **[[https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Documentation|Dokumentationsseite]]** von Asterisk. Alle Neuerungen des aktuellen Release finden sich auf der Seite **[[https://wiki.asterisk.org/wiki/display/AST/New+in+14| New in 14]]**. Die Seite **[[https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Command+Reference|Asterisk 14 Command Reference]]** beinhaltet weitere Unterseiten zu den Kommandos und APIs.
 ===== Installation =====
-Die Installation von Asterisk unetr CentOS 7 könnte man an Hand der **[[http://downloads.asterisk.org/pub/telephony/asterisk/|Programm-Sourcen]]** vornehmen. Dazu müsste auf dem VoIP-Server aber unter anderem Entwicklungstools wie "//**gcc & Co**//" installiert werden; was natürlich auf einem Produktivsystem mehr als zweifelhaft ist und aus Sicherheitsüberlegungen heraus tunlichst vermieden werden sollte.
+Die Installation von Asterisk unetr CentOS 7 könnte man an Hand der **[[https://downloads.asterisk.org/pub/telephony/asterisk/|Programm-Sourcen]]** vornehmen. Dazu müsste auf dem VoIP-Server aber unter anderem Entwicklungstools wie "//**gcc & Co**//" installiert werden; was natürlich auf einem Produktivsystem mehr als zweifelhaft ist und aus Sicherheitsüberlegungen heraus tunlichst vermieden werden sollte.
 ==== Repository ast.tucny.com ====
@@ Zeile 31: / Zeile 34: @@
 **[centosplus]**, **[contrib]** erhält die **priority=2**.
-Den 3rd party repositories, wie z.B. //**[[https://fedoraproject.org/wiki/EPEL|epel]]**// oder //**[[centos:mailserver.guru#priorities_installieren|mailserver.guru]]**// geben wir nun eine Priorität größer-gleich "10", also z.B. **priority=10**.
+Den 3rd party repositories, wie z.B. //**[[https://fedoraproject.org/wiki/EPEL|epel]]**// geben wir nun eine Priorität größer-gleich "10", also z.B. **priority=10**.
 === ast.tucny.com ===
@@ Zeile 118: / Zeile 121: @@
 Repo        : asterisk-14/7/x86_64
 Summary     : The Open Source PBX
-URL         : http://www.asterisk.org/
+URL         : https://www.asterisk.org/
 License     : GPLv2
 Description : Asterisk is a complete PBX in software. It runs on Linux and provides
@@ Zeile 127: / Zeile 130: @@
 ==== RPM-Grundinstallation ====
+=== asterisk ===
 Nachdem unsere Vorbereitungen erst einmal abgeschlossen sind, können wir uns nun daran machen, das Grundpaket **Asterisk** zu installieren.
    # yum install asterisk -y
+Den recht umfangreichen Inhalt des RPM-Paketes können wir uns mit Hilfe des folgenden Aufrufs bei Interesse anzeigen lassen.
+   # rpm -qil asterisk
+<code>Name        : asterisk
+Version     : 14.2.1
+Release     : 1.el7.centos
+Architecture: x86_64
+Install Date: Wed 18 Jan 2017 03:08:52 PM CET
+Group       : Applications/Internet
+Size        : 14393335
+License     : GPLv2
+Signature   : RSA/SHA1, Thu 08 Dec 2016 11:58:05 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         : https://www.asterisk.org/
+Summary     : The Open Source PBX
+Description :
+Asterisk is a complete PBX in software. It runs on Linux and provides
+all of the features you would expect from a PBX and more. Asterisk
+does voice over IP in three protocols, and can interoperate with
+almost all standards-based telephony equipment using relatively
+inexpensive hardware.
+/etc/asterisk
+/etc/asterisk/acl.conf
+/etc/asterisk/adsi.conf
+/etc/asterisk/agents.conf
+/etc/asterisk/alarmreceiver.conf
+/etc/asterisk/amd.conf
+/etc/asterisk/ari.conf
+/etc/asterisk/asterisk.adsi
+/etc/asterisk/asterisk.conf
+/etc/asterisk/ccss.conf
+/etc/asterisk/cdr.conf
+/etc/asterisk/cdr_custom.conf
+/etc/asterisk/cdr_manager.conf
+/etc/asterisk/cdr_syslog.conf
+/etc/asterisk/cel.conf
+/etc/asterisk/cel_custom.conf
+/etc/asterisk/cli.conf
+/etc/asterisk/cli_aliases.conf
+/etc/asterisk/cli_permissions.conf
+/etc/asterisk/codecs.conf
+/etc/asterisk/confbridge.conf
+/etc/asterisk/dnsmgr.conf
+/etc/asterisk/dsp.conf
+/etc/asterisk/dundi.conf
+/etc/asterisk/enum.conf
+/etc/asterisk/extconfig.conf
+/etc/asterisk/extensions.conf
+/etc/asterisk/features.conf
+/etc/asterisk/followme.conf
+/etc/asterisk/http.conf
+/etc/asterisk/indications.conf
+/etc/asterisk/logger.conf
+/etc/asterisk/manager.conf
+/etc/asterisk/modules.conf
+/etc/asterisk/musiconhold.conf
+/etc/asterisk/muted.conf
+/etc/asterisk/osp.conf
+/etc/asterisk/phoneprov.conf
+/etc/asterisk/queuerules.conf
+/etc/asterisk/queues.conf
+/etc/asterisk/res_parking.conf
+/etc/asterisk/res_stun_monitor.conf
+/etc/asterisk/resolver_unbound.conf
+/etc/asterisk/rtp.conf
+/etc/asterisk/say.conf
+/etc/asterisk/sla.conf
+/etc/asterisk/smdi.conf
+/etc/asterisk/sorcery.conf
+/etc/asterisk/stasis.conf
+/etc/asterisk/statsd.conf
+/etc/asterisk/telcordia-1.adsi
+/etc/asterisk/udptl.conf
+/etc/asterisk/users.conf
+/etc/asterisk/vpb.conf
+/etc/logrotate.d/asterisk
+/run/asterisk
+/usr/lib/systemd/system/asterisk.service
+/usr/lib/tmpfiles.d/asterisk.conf
+/usr/lib64/asterisk
+/usr/lib64/asterisk/modules
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+/usr/lib64/asterisk/modules/app_agent_pool.so
+/usr/lib64/asterisk/modules/app_alarmreceiver.so
+/usr/lib64/asterisk/modules/app_amd.so
+/usr/lib64/asterisk/modules/app_authenticate.so
+/usr/lib64/asterisk/modules/app_bridgeaddchan.so
+/usr/lib64/asterisk/modules/app_bridgewait.so
+/usr/lib64/asterisk/modules/app_cdr.so
+/usr/lib64/asterisk/modules/app_celgenuserevent.so
+/usr/lib64/asterisk/modules/app_chanisavail.so
+/usr/lib64/asterisk/modules/app_channelredirect.so
+/usr/lib64/asterisk/modules/app_chanspy.so
+/usr/lib64/asterisk/modules/app_confbridge.so
+/usr/lib64/asterisk/modules/app_controlplayback.so
+/usr/lib64/asterisk/modules/app_db.so
+/usr/lib64/asterisk/modules/app_dial.so
+/usr/lib64/asterisk/modules/app_dictate.so
+/usr/lib64/asterisk/modules/app_directed_pickup.so
+/usr/lib64/asterisk/modules/app_disa.so
+/usr/lib64/asterisk/modules/app_dumpchan.so
+/usr/lib64/asterisk/modules/app_echo.so
+/usr/lib64/asterisk/modules/app_exec.so
+/usr/lib64/asterisk/modules/app_externalivr.so
+/usr/lib64/asterisk/modules/app_followme.so
+/usr/lib64/asterisk/modules/app_forkcdr.so
+/usr/lib64/asterisk/modules/app_getcpeid.so
+/usr/lib64/asterisk/modules/app_image.so
+/usr/lib64/asterisk/modules/app_macro.so
+/usr/lib64/asterisk/modules/app_milliwatt.so
+/usr/lib64/asterisk/modules/app_mixmonitor.so
+/usr/lib64/asterisk/modules/app_morsecode.so
+/usr/lib64/asterisk/modules/app_nbscat.so
+/usr/lib64/asterisk/modules/app_originate.so
+/usr/lib64/asterisk/modules/app_playback.so
+/usr/lib64/asterisk/modules/app_playtones.so
+/usr/lib64/asterisk/modules/app_privacy.so
+/usr/lib64/asterisk/modules/app_queue.so
+/usr/lib64/asterisk/modules/app_read.so
+/usr/lib64/asterisk/modules/app_readexten.so
+/usr/lib64/asterisk/modules/app_record.so
+/usr/lib64/asterisk/modules/app_saycounted.so
+/usr/lib64/asterisk/modules/app_sayunixtime.so
+/usr/lib64/asterisk/modules/app_senddtmf.so
+/usr/lib64/asterisk/modules/app_sendtext.so
+/usr/lib64/asterisk/modules/app_setcallerid.so
+/usr/lib64/asterisk/modules/app_sms.so
+/usr/lib64/asterisk/modules/app_softhangup.so
+/usr/lib64/asterisk/modules/app_speech_utils.so
+/usr/lib64/asterisk/modules/app_stack.so
+/usr/lib64/asterisk/modules/app_stasis.so
+/usr/lib64/asterisk/modules/app_statsd.so
+/usr/lib64/asterisk/modules/app_system.so
+/usr/lib64/asterisk/modules/app_talkdetect.so
+/usr/lib64/asterisk/modules/app_test.so
+/usr/lib64/asterisk/modules/app_transfer.so
+/usr/lib64/asterisk/modules/app_url.so
+/usr/lib64/asterisk/modules/app_userevent.so
+/usr/lib64/asterisk/modules/app_verbose.so
+/usr/lib64/asterisk/modules/app_waitforring.so
+/usr/lib64/asterisk/modules/app_waitforsilence.so
+/usr/lib64/asterisk/modules/app_waituntil.so
+/usr/lib64/asterisk/modules/app_while.so
+/usr/lib64/asterisk/modules/app_zapateller.so
+/usr/lib64/asterisk/modules/bridge_builtin_features.so
+/usr/lib64/asterisk/modules/bridge_builtin_interval_features.so
+/usr/lib64/asterisk/modules/bridge_holding.so
+/usr/lib64/asterisk/modules/bridge_native_rtp.so
+/usr/lib64/asterisk/modules/bridge_simple.so
+/usr/lib64/asterisk/modules/bridge_softmix.so
+/usr/lib64/asterisk/modules/cdr_csv.so
+/usr/lib64/asterisk/modules/cdr_custom.so
+/usr/lib64/asterisk/modules/cdr_manager.so
+/usr/lib64/asterisk/modules/cdr_syslog.so
+/usr/lib64/asterisk/modules/cel_custom.so
+/usr/lib64/asterisk/modules/cel_manager.so
+/usr/lib64/asterisk/modules/chan_bridge_media.so
+/usr/lib64/asterisk/modules/chan_rtp.so
+/usr/lib64/asterisk/modules/codec_a_mu.so
+/usr/lib64/asterisk/modules/codec_adpcm.so
+/usr/lib64/asterisk/modules/codec_alaw.so
+/usr/lib64/asterisk/modules/codec_g722.so
+/usr/lib64/asterisk/modules/codec_g726.so
+/usr/lib64/asterisk/modules/codec_gsm.so
+/usr/lib64/asterisk/modules/codec_ilbc.so
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+/usr/lib64/asterisk/modules/codec_resample.so
+/usr/lib64/asterisk/modules/codec_speex.so
+/usr/lib64/asterisk/modules/codec_ulaw.so
+/usr/lib64/asterisk/modules/format_g719.so
+/usr/lib64/asterisk/modules/format_g723.so
+/usr/lib64/asterisk/modules/format_g726.so
+/usr/lib64/asterisk/modules/format_g729.so
+/usr/lib64/asterisk/modules/format_gsm.so
+/usr/lib64/asterisk/modules/format_h263.so
+/usr/lib64/asterisk/modules/format_h264.so
+/usr/lib64/asterisk/modules/format_ilbc.so
+/usr/lib64/asterisk/modules/format_jpeg.so
+/usr/lib64/asterisk/modules/format_ogg_speex.so
+/usr/lib64/asterisk/modules/format_ogg_vorbis.so
+/usr/lib64/asterisk/modules/format_pcm.so
+/usr/lib64/asterisk/modules/format_siren14.so
+/usr/lib64/asterisk/modules/format_siren7.so
+/usr/lib64/asterisk/modules/format_sln.so
+/usr/lib64/asterisk/modules/format_vox.so
+/usr/lib64/asterisk/modules/format_wav.so
+/usr/lib64/asterisk/modules/format_wav_gsm.so
+/usr/lib64/asterisk/modules/func_aes.so
+/usr/lib64/asterisk/modules/func_audiohookinherit.so
+/usr/lib64/asterisk/modules/func_base64.so
+/usr/lib64/asterisk/modules/func_blacklist.so
+/usr/lib64/asterisk/modules/func_callcompletion.so
+/usr/lib64/asterisk/modules/func_callerid.so
+/usr/lib64/asterisk/modules/func_cdr.so
+/usr/lib64/asterisk/modules/func_channel.so
+/usr/lib64/asterisk/modules/func_config.so
+/usr/lib64/asterisk/modules/func_cut.so
+/usr/lib64/asterisk/modules/func_db.so
+/usr/lib64/asterisk/modules/func_devstate.so
+/usr/lib64/asterisk/modules/func_dialgroup.so
+/usr/lib64/asterisk/modules/func_dialplan.so
+/usr/lib64/asterisk/modules/func_enum.so
+/usr/lib64/asterisk/modules/func_env.so
+/usr/lib64/asterisk/modules/func_extstate.so
+/usr/lib64/asterisk/modules/func_frame_trace.so
+/usr/lib64/asterisk/modules/func_global.so
+/usr/lib64/asterisk/modules/func_groupcount.so
+/usr/lib64/asterisk/modules/func_hangupcause.so
+/usr/lib64/asterisk/modules/func_holdintercept.so
+/usr/lib64/asterisk/modules/func_iconv.so
+/usr/lib64/asterisk/modules/func_jitterbuffer.so
+/usr/lib64/asterisk/modules/func_lock.so
+/usr/lib64/asterisk/modules/func_logic.so
+/usr/lib64/asterisk/modules/func_math.so
+/usr/lib64/asterisk/modules/func_md5.so
+/usr/lib64/asterisk/modules/func_module.so
+/usr/lib64/asterisk/modules/func_periodic_hook.so
+/usr/lib64/asterisk/modules/func_pitchshift.so
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+/usr/lib64/asterisk/modules/func_shell.so
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+/usr/lib64/asterisk/modules/func_sprintf.so
+/usr/lib64/asterisk/modules/func_srv.so
+/usr/lib64/asterisk/modules/func_strings.so
+/usr/lib64/asterisk/modules/func_sysinfo.so
+/usr/lib64/asterisk/modules/func_talkdetect.so
+/usr/lib64/asterisk/modules/func_timeout.so
+/usr/lib64/asterisk/modules/func_uri.so
+/usr/lib64/asterisk/modules/func_version.so
+/usr/lib64/asterisk/modules/func_volume.so
+/usr/lib64/asterisk/modules/pbx_config.so
+/usr/lib64/asterisk/modules/pbx_dundi.so
+/usr/lib64/asterisk/modules/pbx_loopback.so
+/usr/lib64/asterisk/modules/pbx_realtime.so
+/usr/lib64/asterisk/modules/pbx_spool.so
+/usr/lib64/asterisk/modules/res_adsi.so
+/usr/lib64/asterisk/modules/res_agi.so
+/usr/lib64/asterisk/modules/res_ari.so
+/usr/lib64/asterisk/modules/res_ari_applications.so
+/usr/lib64/asterisk/modules/res_ari_asterisk.so
+/usr/lib64/asterisk/modules/res_ari_bridges.so
+/usr/lib64/asterisk/modules/res_ari_channels.so
+/usr/lib64/asterisk/modules/res_ari_device_states.so
+/usr/lib64/asterisk/modules/res_ari_endpoints.so
+/usr/lib64/asterisk/modules/res_ari_events.so
+/usr/lib64/asterisk/modules/res_ari_model.so
+/usr/lib64/asterisk/modules/res_ari_playbacks.so
+/usr/lib64/asterisk/modules/res_ari_recordings.so
+/usr/lib64/asterisk/modules/res_ari_sounds.so
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+/usr/lib64/asterisk/modules/res_crypto.so
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+/usr/lib64/asterisk/modules/res_format_attr_h263.so
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+/usr/lib64/asterisk/modules/res_format_attr_ilbc.so
+/usr/lib64/asterisk/modules/res_format_attr_opus.so
+/usr/lib64/asterisk/modules/res_format_attr_silk.so
+/usr/lib64/asterisk/modules/res_format_attr_siren14.so
+/usr/lib64/asterisk/modules/res_format_attr_siren7.so
+/usr/lib64/asterisk/modules/res_format_attr_vp8.so
+/usr/lib64/asterisk/modules/res_http_media_cache.so
+/usr/lib64/asterisk/modules/res_http_post.so
+/usr/lib64/asterisk/modules/res_http_websocket.so
+/usr/lib64/asterisk/modules/res_limit.so
+/usr/lib64/asterisk/modules/res_manager_devicestate.so
+/usr/lib64/asterisk/modules/res_manager_presencestate.so
+/usr/lib64/asterisk/modules/res_monitor.so
+/usr/lib64/asterisk/modules/res_musiconhold.so
+/usr/lib64/asterisk/modules/res_mutestream.so
+/usr/lib64/asterisk/modules/res_parking.so
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+/usr/lib64/asterisk/modules/res_realtime.so
+/usr/lib64/asterisk/modules/res_rtp_asterisk.so
+/usr/lib64/asterisk/modules/res_rtp_multicast.so
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+/usr/lib64/asterisk/modules/res_smdi.so
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+/usr/lib64/asterisk/modules/res_stasis.so
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+/usr/sbin/astcanary
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+/usr/sbin/astman
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+/usr/sbin/autosupport
+/usr/sbin/check_expr
+/usr/sbin/check_expr2
+/usr/sbin/muted
+/usr/sbin/rasterisk
+/usr/sbin/smsq
+/usr/sbin/stereorize
+/usr/sbin/streamplayer
+/usr/share/asterisk
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+/usr/share/asterisk/documentation
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+/usr/share/asterisk/documentation/thirdparty
+/usr/share/asterisk/images
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+/usr/share/asterisk/images/kpad2.jpg
+/usr/share/asterisk/keys
+/usr/share/asterisk/moh
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+/usr/share/asterisk/rest-api/asterisk.json
+/usr/share/asterisk/rest-api/bridges.json
+/usr/share/asterisk/rest-api/channels.json
+/usr/share/asterisk/rest-api/deviceStates.json
+/usr/share/asterisk/rest-api/endpoints.json
+/usr/share/asterisk/rest-api/events.json
+/usr/share/asterisk/rest-api/mailboxes.json
+/usr/share/asterisk/rest-api/playbacks.json
+/usr/share/asterisk/rest-api/recordings.json
+/usr/share/asterisk/rest-api/resources.json
+/usr/share/asterisk/rest-api/sounds.json
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+/usr/share/doc/asterisk-14.2.1/ChangeLog
+/usr/share/doc/asterisk-14.2.1/README
+/usr/share/doc/asterisk-14.2.1/README-SERIOUSLY.bestpractices.txt
+/usr/share/doc/asterisk-14.2.1/README-addons.txt
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+/usr/share/doc/asterisk-14.2.1/UPGRADE-1.8.txt
+/usr/share/doc/asterisk-14.2.1/UPGRADE-10.txt
+/usr/share/doc/asterisk-14.2.1/UPGRADE-11.txt
+/usr/share/doc/asterisk-14.2.1/UPGRADE-12.txt
+/usr/share/doc/asterisk-14.2.1/UPGRADE-13.txt
+/usr/share/doc/asterisk-14.2.1/UPGRADE.txt
+/usr/share/doc/asterisk-14.2.1/Zaptel-to-DAHDI.txt
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+/usr/share/doc/asterisk-14.2.1/configs
+/usr/share/doc/asterisk-14.2.1/configs/basic-pbx
+/usr/share/doc/asterisk-14.2.1/configs/basic-pbx/README
+/usr/share/doc/asterisk-14.2.1/configs/basic-pbx/asterisk.conf
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+/usr/share/doc/asterisk-14.2.1/configs/basic-pbx/modules.conf
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+/usr/share/doc/asterisk-14.2.1/configs/basic-pbx/pjsip.conf
+/usr/share/doc/asterisk-14.2.1/configs/basic-pbx/voicemail.conf
+/usr/share/doc/asterisk-14.2.1/configs/samples
+/usr/share/doc/asterisk-14.2.1/configs/samples/acl.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/adsi.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/agents.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/alarmreceiver.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/alsa.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/amd.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/app_mysql.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/app_skel.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/ari.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/asterisk.adsi
+/usr/share/doc/asterisk-14.2.1/configs/samples/asterisk.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/calendar.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/ccss.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr_adaptive_odbc.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr_custom.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr_manager.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr_mysql.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr_odbc.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr_pgsql.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr_sqlite3_custom.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr_syslog.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cdr_tds.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cel.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cel_custom.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cel_odbc.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cel_pgsql.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cel_sqlite3_custom.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cel_tds.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/chan_dahdi.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/chan_mobile.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cli.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cli_aliases.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/cli_permissions.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/codecs.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/confbridge.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/config_test.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/console.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/dbsep.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/dnsmgr.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/dsp.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/dundi.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/enum.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/extconfig.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/extensions.ael.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/extensions.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/extensions.lua.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/extensions_minivm.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/features.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/festival.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/followme.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/func_odbc.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/hep.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/http.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/iax.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/iaxprov.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/indications.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/logger.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/manager.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/meetme.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/mgcp.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/minivm.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/misdn.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/modules.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/motif.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/musiconhold.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/muted.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/ooh323.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/osp.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/oss.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/phone.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/phoneprov.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/pjproject.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/pjsip.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/pjsip_notify.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/pjsip_wizard.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/queuerules.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/queues.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_config_mysql.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_config_sqlite.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_config_sqlite3.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_corosync.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_curl.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_fax.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_ldap.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_odbc.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_parking.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_pgsql.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_pktccops.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_snmp.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/res_stun_monitor.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/resolver_unbound.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/rtp.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/say.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/sip.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/sip_notify.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/skinny.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/sla.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/smdi.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/sorcery.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/ss7.timers.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/stasis.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/statsd.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/telcordia-1.adsi
+/usr/share/doc/asterisk-14.2.1/configs/samples/test_sorcery.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/udptl.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/unistim.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/users.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/voicemail.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/vpb.conf.sample
+/usr/share/doc/asterisk-14.2.1/configs/samples/xmpp.conf.sample
+/usr/share/man/man8/astdb2bdb.8.gz
+/usr/share/man/man8/astdb2sqlite3.8.gz
+/usr/share/man/man8/asterisk.8.gz
+/usr/share/man/man8/astgenkey.8.gz
+/usr/share/man/man8/autosupport.8.gz
+/usr/share/man/man8/safe_asterisk.8.gz
+/var/lib/asterisk
+/var/log/asterisk
+/var/log/asterisk/cdr-csv
+/var/log/asterisk/cdr-custom
+/var/spool/asterisk
+/var/spool/asterisk/monitor
+/var/spool/asterisk/outgoing
+/var/spool/asterisk/tmp
+/var/spool/asterisk/uploads
+/var/spool/asterisk/voicemail</code>
+=== asterisk-iax2 ===
+Da wir später z.B. für die Clients auf das **I**nter**A**sterisk e**X**change Protokoll oder kurz **[[https://de.wikipedia.org/wiki/InterAsterisk_eXchange|IAX2]]** zurückgreifen wollen, installieren wir hierzu das zugehörige **RPM**.
+   # yum install asterisk-iax2
+Was genau diese RPM-Paket in unserern Server mitgebracht hat zeigt uns wie immer der Aufruf des Befehls **rpm** mit der Option **-qil**.
+   # rpm -qil asterisk-iax2
+<code>Name        : asterisk-iax2
+Version     : 14.2.1
+Release     : 1.el7.centos
+Architecture: x86_64
+Install Date: Wed 18 Jan 2017 07:08:23 PM CET
+Group       : Applications/Internet
+Size        : 389194
+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         : https://www.asterisk.org/
+Summary     : IAX2 channel driver for Asterisk
+Description :
+IAX2 channel driver for Asterisk
+/etc/asterisk/iax.conf
+/etc/asterisk/iaxprov.conf
+/usr/lib64/asterisk/modules/chan_iax2.so
+/usr/share/asterisk/firmware
+/usr/share/asterisk/firmware/iax</code>
+=== asterisk-pjsip ===
+Für die Anbindung unseres Asterisk-Servers an den Provider nutzen wir das **S**ession **I**nitiation **P**rotocol - **[[https://de.wikipedia.org/wiki/Session_Initiation_Protocol|SIP]]**. Statt dem alten chan_sip Modulen benutzen wir unter //**Asterisk 14**// nunmehr die neuen, auf den **[[https://www.pjsip.org/about.htm|PJSIP-Bibliotheken]]** basierenden, SIP-Stack. die wesentlichen Unterschiede bei der Konfiguration sind auf dieser **[[https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip|Seite]]** ausführlich beschrieben.
+Wie installieren also auch hier das passende **RPM**.
+   # yum install asterisk-pjsip
+Was genau diese RPM-Paket in unseren Server mitgebracht hat zeigt uns wie immer der Aufruf des Befehls **rpm** mit der Option **-qil**.
+   # rpm -qil asterisk-pjsip
+<code>Name        : asterisk-pjsip
+Version     : 14.2.1
+Release     : 1.el7.centos
+Architecture: x86_64
+Install Date: Thu 19 Jan 2017 02:21:55 PM CET
+Group       : Applications/Internet
+Size        : 1554633
+License     : GPLv2
+Signature   : RSA/SHA1, Thu 08 Dec 2016 11:58:06 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         : https://www.asterisk.org/
+Summary     : SIP channel based upon the PJSIP library
+Description :
+SIP channel based upon the PJSIP library
+/etc/asterisk/pjproject.conf
+/etc/asterisk/pjsip.conf
+/etc/asterisk/pjsip_notify.conf
+/etc/asterisk/pjsip_wizard.conf
+/usr/lib64/asterisk/modules/chan_pjsip.so
+/usr/lib64/asterisk/modules/func_pjsip_aor.so
+/usr/lib64/asterisk/modules/func_pjsip_contact.so
+/usr/lib64/asterisk/modules/func_pjsip_endpoint.so
+/usr/lib64/asterisk/modules/res_pjproject.so
+/usr/lib64/asterisk/modules/res_pjsip.so
+/usr/lib64/asterisk/modules/res_pjsip_acl.so
+/usr/lib64/asterisk/modules/res_pjsip_authenticator_digest.so
+/usr/lib64/asterisk/modules/res_pjsip_caller_id.so
+/usr/lib64/asterisk/modules/res_pjsip_config_wizard.so
+/usr/lib64/asterisk/modules/res_pjsip_dialog_info_body_generator.so
+/usr/lib64/asterisk/modules/res_pjsip_diversion.so
+/usr/lib64/asterisk/modules/res_pjsip_dlg_options.so
+/usr/lib64/asterisk/modules/res_pjsip_dtmf_info.so
+/usr/lib64/asterisk/modules/res_pjsip_empty_info.so
+/usr/lib64/asterisk/modules/res_pjsip_endpoint_identifier_anonymous.so
+/usr/lib64/asterisk/modules/res_pjsip_endpoint_identifier_ip.so
+/usr/lib64/asterisk/modules/res_pjsip_endpoint_identifier_user.so
+/usr/lib64/asterisk/modules/res_pjsip_exten_state.so
+/usr/lib64/asterisk/modules/res_pjsip_header_funcs.so
+/usr/lib64/asterisk/modules/res_pjsip_history.so
+/usr/lib64/asterisk/modules/res_pjsip_logger.so
+/usr/lib64/asterisk/modules/res_pjsip_messaging.so
+/usr/lib64/asterisk/modules/res_pjsip_mwi.so
+/usr/lib64/asterisk/modules/res_pjsip_mwi_body_generator.so
+/usr/lib64/asterisk/modules/res_pjsip_nat.so
+/usr/lib64/asterisk/modules/res_pjsip_notify.so
+/usr/lib64/asterisk/modules/res_pjsip_one_touch_record_info.so
+/usr/lib64/asterisk/modules/res_pjsip_outbound_authenticator_digest.so
+/usr/lib64/asterisk/modules/res_pjsip_outbound_publish.so
+/usr/lib64/asterisk/modules/res_pjsip_outbound_registration.so
+/usr/lib64/asterisk/modules/res_pjsip_path.so
+/usr/lib64/asterisk/modules/res_pjsip_phoneprov_provider.so
+/usr/lib64/asterisk/modules/res_pjsip_pidf_body_generator.so
+/usr/lib64/asterisk/modules/res_pjsip_pidf_digium_body_supplement.so
+/usr/lib64/asterisk/modules/res_pjsip_pidf_eyebeam_body_supplement.so
+/usr/lib64/asterisk/modules/res_pjsip_publish_asterisk.so
+/usr/lib64/asterisk/modules/res_pjsip_pubsub.so
+/usr/lib64/asterisk/modules/res_pjsip_refer.so
+/usr/lib64/asterisk/modules/res_pjsip_registrar.so
+/usr/lib64/asterisk/modules/res_pjsip_registrar_expire.so
+/usr/lib64/asterisk/modules/res_pjsip_rfc3326.so
+/usr/lib64/asterisk/modules/res_pjsip_sdp_rtp.so
+/usr/lib64/asterisk/modules/res_pjsip_send_to_voicemail.so
+/usr/lib64/asterisk/modules/res_pjsip_session.so
+/usr/lib64/asterisk/modules/res_pjsip_sips_contact.so
+/usr/lib64/asterisk/modules/res_pjsip_t38.so
+/usr/lib64/asterisk/modules/res_pjsip_transport_management.so
+/usr/lib64/asterisk/modules/res_pjsip_transport_websocket.so
+/usr/lib64/asterisk/modules/res_pjsip_xpidf_body_generator.so</code>
+===== Programmstart =====
+==== erster manueller Start ====
+Abwohl wie noch keine individuellen Konfigurationsänderungen vorgenommen haben, können wir den Asterisk-Daemon schon starten. Hierzu nutzen wir das **systemd** Startscript.
+   # systemctl start asterisk.service
+Den erfolgreichen Start des Daemon können wir nun abfragen.
+   # systemctl status asterisk.service -l
+<html><pre class="code">
+<font style="color: rgb(0, 255, 0)"><b>● </b></font><font style="color: rgb(0, 0, 0)">asterisk.service - Asterisk PBX and telephony daemon.
+   Loaded: loaded (/usr/lib/systemd/system/asterisk.service; disabled; vendor preset: disabled)
+   Active: <font style="color: rgb(0, 255, 0)"><b>active (running) </b></font><font style="color: rgb(0, 0, 0)"> since Wed 2017-01-18 19:27:13 CET; 29s ago
+ Main PID: 12988 (asterisk)
+   CGroup: /system.slice/asterisk.service
+           └─12988 /usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf
+Jan 18 19:27:14 vml000177.dmz.nausch.org asterisk[12988]: [Jan 18 19:27:14] WARNING[12988]: res_phoneprov.c:1231 get_defaults: Unable to find a valid server address or name.
+Jan 18 19:27:14 vml000177.dmz.nausch.org asterisk[12988]: [Jan 18 19:27:14] ERROR[12988]: ari/config.c:314 process_config: No configured users for ARI
+Jan 18 19:27:14 vml000177.dmz.nausch.org asterisk[12988]: 19:27:14.532 os_core_unix.c !pjlib 2.3 for POSIX initialized
+Jan 18 19:27:14 vml000177.dmz.nausch.org asterisk[12988]: [Jan 18 19:27:14] WARNING[12988]: res_musiconhold.c:1767 load_moh_classes: A directory must be specified for class 'general'!
+Jan 18 19:27:14 vml000177.dmz.nausch.org asterisk[12988]: [Jan 18 19:27:14] WARNING[12988]: res_musiconhold.c:1945 load_module: No music on hold classes configured, disabling music on hold.
+Jan 18 19:27:14 vml000177.dmz.nausch.org asterisk[12988]: 19:27:14.539          pjlib !select() I/O Queue created (0x7f33ac000948)
+Jan 18 19:27:14 vml000177.dmz.nausch.org asterisk[12988]: SIP channel loading...
+Jan 18 19:27:14 vml000177.dmz.nausch.org asterisk[12988]: [Jan 18 19:27:14] NOTICE[12988]: confbridge/conf_config_parser.c:2126 verify_default_profiles: Adding default_menu menu to app_confbridge
+Jan 18 19:27:14 vml000177.dmz.nausch.org asterisk[12988]: [Jan 18 19:27:14] NOTICE[12988]: cel_custom.c:97 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
+Jan 18 19:27:15 vml000177.dmz.nausch.org asterisk[12988]: Asterisk Ready.</font>
+</pre></html>
+Im Verzeichnis //**/var/log/asterisk**// beschreibt der Daemon auch ein eigene Logdatei
+   # less /var/log/asterisk/messages
+<code>[Jan 18 19:27:13] Asterisk 14.2.1 built by mockbuild @ olive.ph.tucny.com on a x86_64 running Linux on 2016-12-08 22:53:51 UTC
+[Jan 18 19:27:14] NOTICE[12988] cdr.c: CDR simple logging enabled.
+[Jan 18 19:27:14] NOTICE[12988] loader.c: 223 modules will be loaded.
+[Jan 18 19:27:14] WARNING[12988] res_phoneprov.c: Unable to find a valid server address or name.
+[Jan 18 19:27:14] ERROR[12988] ari/config.c: No configured users for ARI
+[Jan 18 19:27:14] WARNING[12988] res_musiconhold.c: A directory must be specified for class 'general'!
+[Jan 18 19:27:14] WARNING[12988] res_musiconhold.c: No music on hold classes configured, disabling music on hold.
+[Jan 18 19:27:14] NOTICE[12988] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
+[Jan 18 19:27:14] NOTICE[12988] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.</code>
+==== automatischer Start ====
+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
+===== Asterisk-Konsole =====
+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
+<code>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></code>
+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
+<code>!                              -- 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</code>
+Eine Übersicht der wesentlichen Einstellungsoptionen können wir uns z.B. mit dem Befehl ''core show settings'' anzeigen lassen.
+  vml000177*CLI> core show settings
+<code>
+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
+</code>
+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
+===== Paketfilter =====
+   # firewall-cmd --get-active-zones
+<code>public
+  interfaces: eth0 eth2 eth1</code>
+   # 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
+<code>internal
+  interfaces: eth1
+public
+  interfaces: eth0 eth2</code>
+   # 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
+<code>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
+</code>
+   # firewall-cmd --get-active-zones
+<code>internal
+  interfaces: eth1
+public
+  interfaces: eth0 eth2</code>
+   # 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
+<code>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</code>
+===== iax2 =====
+   # asterisk -r
+<code>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></code>
+  vml000177*CLI> iax2 show peers
+<code>Name/Username    Host                                           Mask                                      Port           Status      Description
+/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
+iax2 peers [0 online, 0 offline, 2 unmonitored]</code>
+===== dialplan =====
+   # asterisk -r
+<code>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></code>
+  vml000177*CLI> dialplan show
+<code>[ Context '__func_periodic_hook_context__' created by 'func_periodic_hook' ]
+  'beep' (CID match '') =>  1. Answer()                                   [func_periodic_hook]
+. Playback(beep)                             [func_periodic_hook]
+  'hook' (CID match '') =>  1. Set(EncodedChannel=${CUT(HOOK_CHANNEL,-,1-2)}) [func_periodic_hook]
+. Set(GROUP_NAME=${EncodedChannel}${HOOK_ID}) [func_periodic_hook]
+. Set(GROUP(periodic-hook)=${GROUP_NAME})    [func_periodic_hook]
+. ExecIf($[${GROUP_COUNT(${GROUP_NAME}@periodic-hook)} > 1]?Hangup()) [func_periodic_hook]
+. Set(ChannelToSpy=${URIDECODE(${EncodedChannel})}) [func_periodic_hook]
+. ChanSpy(${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: ${EXTEN})                    [pbx_config]
+. Wait(0.25)                             [pbx_config]
+. Answer()                               [pbx_config]
+. Playback(vm-from)                      [pbx_config]
+. SayDigits(${CALLERID(ani)})            [pbx_config]
+. Wait(1.25)                             [pbx_config]
+. SayDigits(${CALLERID(ani)})            [pbx_config]
+. Return()                               [pbx_config]
+[ Context 'time' created by 'pbx_config' ]
+  '_X.' =>          30000. NoOp(Time: ${EXTEN} ${timezone})       [pbx_config]
+. Wait(0.25)                             [pbx_config]
+. Answer()                               [pbx_config]
+. Set(FUTURETIME=$[${EPOCH} + 12])       [pbx_config]
+. SayUnixTime(${FUTURETIME},Zulu,HNS)    [pbx_config]
+. SayPhonetic(z)                         [pbx_config]
+. SayUnixTime(${FUTURETIME},${timezone},HNS) [pbx_config]
+. Playback(spy-local)                    [pbx_config]
+. WaitUntil(${FUTURETIME})               [pbx_config]
+. Playback(beep)                         [pbx_config]
+. Return()                               [pbx_config]
+[ Context 'default' created by 'pbx_config' ]
+  Include =>        'demo'                                        [pbx_config]
+[ Context 'public' created by 'pbx_config' ]
+  Include =>        'demo'                                        [pbx_config]
+[ Context 'page' created by 'pbx_config' ]
+  '_X.' =>          1. Macro(page,SIP/${EXTEN})                   [pbx_config]
+[ Context 'demo' created by 'pbx_config' ]
+  '#' =>            1. Playback(demo-thanks)                      [pbx_config]
+. Hangup()                                   [pbx_config]
+  '1000' =>         1. Goto(default,s,1)                          [pbx_config]
+  '1234' =>         1. Playback(transfer,skip)                    [pbx_config]
+. Gosub(${EXTEN},stdexten(${GLOBAL(CONSOLE)})) [pbx_config]
+. Goto(default,s,1)                          [pbx_config]
+  '1235' =>         1. Voicemail(1234,u)                          [pbx_config]
+  '1236' =>         1. Dial(Console/dsp)                          [pbx_config]
+. Voicemail(1234,b)                          [pbx_config]
+  '2' =>            1. BackGround(demo-moreinfo)                  [pbx_config]
+. Goto(s,instruct)                           [pbx_config]
+  '3' =>            1. Set(CHANNEL(language)=fr)                  [pbx_config]
+. Goto(s,restart)                            [pbx_config]
+  '500' =>          1. Playback(demo-abouttotry)                  [pbx_config]
+. Dial(IAX2/guest@pbx.digium.com/s@default)  [pbx_config]
+. Playback(demo-nogo)                        [pbx_config]
+. Goto(s,6)                                  [pbx_config]
+  '600' =>          1. Playback(demo-echotest)                    [pbx_config]
+. Echo()                                     [pbx_config]
+. Playback(demo-echodone)                    [pbx_config]
+. Goto(s,6)                                  [pbx_config]
+  '76245' =>        1. Macro(page,SIP/Grandstream1)               [pbx_config]
+  '7999' =>         1. Set(TIMEOUT(absolute)=60)                  [pbx_config]
+. Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n,d) [pbx_config]
+  '8500' =>         1. VoicemailMain()                            [pbx_config]
+. Goto(s,6)                                  [pbx_config]
+  'i' =>            1. Playback(invalid)                          [pbx_config]
+  's' =>            1. Wait(1)                                    [pbx_config]
+. Answer()                                   [pbx_config]
+. Set(TIMEOUT(digit)=5)                      [pbx_config]
+. Set(TIMEOUT(response)=10)                  [pbx_config]
+     [restart]      5. BackGround(demo-congrats)                  [pbx_config]
+     [instruct]     6. BackGround(demo-instruct)                  [pbx_config]
+. 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]
+. GoToIf($[${AVAILSTATUS} = "1"]?autoanswer:fail) [pbx_config]
+     [autoanswer]   3. Set(_ALERT_INFO="RA")                      [pbx_config]
+. SIPAddHeader(Call-Info: Answer-After=0)    [pbx_config]
+. NoOp()                                     [pbx_config]
+. Dial(${ARG1})                              [pbx_config]
+     [fail]         7. Hangup()                                   [pbx_config]
+[ Context 'stdPrivacyexten' created by 'pbx_config' ]
+  'a' =>            1. VoicemailMain(${mbx})                      [pbx_config]
+. Return()                                   [pbx_config]
+  'stdexten-BUSY' => 1. Voicemail(${mbx},b)                        [pbx_config]
+. NoOp(Finish stdPrivacyexten BUSY)          [pbx_config]
+. Return()                                   [pbx_config]
+  'stdexten-DONTCALL' => 1. Goto(${dontcntx},s,1)                      [pbx_config]
+  'stdexten-NOANSWER' => 1. Voicemail(${mbx},u)                        [pbx_config]
+. NoOp(Finish stdPrivacyexten NOANSWER)      [pbx_config]
+. Return()                                   [pbx_config]
+  'stdexten-TORTURE' => 1. Goto(${tortcntx},s,1)                      [pbx_config]
+  '_stde[x]te[n]-.' => 1. Goto(stdexten-NOANSWER,1)                  [pbx_config]
+  '_X.' =>          60000. NoOp(Start stdPrivacyexten)            [pbx_config]
+. Set(LOCAL(ext)=${ARG1})                [pbx_config]
+. Set(LOCAL(dev)=${ARG2})                [pbx_config]
+. Set(LOCAL(dontcntx)=${ARG3})           [pbx_config]
+. Set(LOCAL(tortcntx)=${ARG4})           [pbx_config]
+. Set(LOCAL(cntx)=${ARG5})               [pbx_config]
+. Set(LOCAL(mbx)="${ext}"$["${cntx}" ? "@${cntx}" :: ""]) [pbx_config]
+. Dial(${dev},20,p)                      [pbx_config]
+. Goto(stdexten-${DIALSTATUS},1)         [pbx_config]
+[ Context 'stdexten' created by 'pbx_config' ]
+  'a' =>            1. VoicemailMain(${mbx})                      [pbx_config]
+. Return()                                   [pbx_config]
+  'stdexten-BUSY' => 1. Voicemail(${mbx},b)                        [pbx_config]
+. Return()                                   [pbx_config]
+  'stdexten-NOANSWER' => 1. Voicemail(${mbx},u)                        [pbx_config]
+. Return()                                   [pbx_config]
+  '_stde[x]te[n]-.' => 1. Goto(stdexten-NOANSWER,1)                  [pbx_config]
+  '_X.' =>          50000. NoOp(Start stdexten)                   [pbx_config]
+. Set(LOCAL(ext)=${EXTEN})               [pbx_config]
+. Set(LOCAL(dev)=${ARG1})                [pbx_config]
+. Set(LOCAL(cntx)=${ARG2})               [pbx_config]
+. Set(LOCAL(mbx)=${ext}${IF($[!${ISNULL(${cntx})}]?@${cntx})}) [pbx_config]
+. Dial(${dev},20)                        [pbx_config]
+. Goto(stdexten-${DIALSTATUS},1)         [pbx_config]
+[ Context 'macro-trunkdial' created by 'pbx_config' ]
+  's' =>            1. Dial(${ARG1})                              [pbx_config]
+. 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]
+. Congestion()                               [pbx_config]
+  '_X!' =>          1. Verbose(2,Performing ISN lookup for ${EXTEN}) [pbx_config]
+. Set(SUFFIX=${CUT(EXTEN,*,2-)})             [pbx_config]
+. GotoIf($["${FILTER(0-9,${SUFFIX})}" != "${SUFFIX}"]?fn-CONGESTION,1) [pbx_config]
+. Set(TIMEOUT(absolute)=10800)               [pbx_config]
+. Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)}) [pbx_config]
+. GotoIf($["${isnresult}" != ""]?from)       [pbx_config]
+. Set(DIALSTATUS=CONGESTION)                 [pbx_config]
+. Goto(fn-CONGESTION,1)                      [pbx_config]
+     [from]         9. Set(__SIPFROMUSER=${CALLERID(num)})        [pbx_config]
+. GotoIf($["${GLOBAL(FREENUMDOMAIN)}" = ""]?dial) [pbx_config]
+. Set(__SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)}) [pbx_config]
+     [dial]         12. Dial(SIP/${isnresult},40)                 [pbx_config]
+. Goto(fn-${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]
+. Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [pbx_config]
+[ Context 'trunkint' created by 'pbx_config' ]
+  '_9011.' =>       1. Macro(dundi-e164,${EXTEN:4})               [pbx_config]
+. 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. =-</code>
+==== konsole verbose output ====
+  vml000177*CLI> core set verbose 5
+  Console verbose was OFF and is now 5.
+==== echotest ====
+<code>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'</code>
+==== demo ====
+<code>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'</code>
+==== Beispiele ====
+=== dial ===
+https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Dial
+   # vim /etc/asterisk/extensions.conf
+<code>; 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)
+</code>
+   # asterisk -vvvvr
+<code>    -- 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'
+</code>
+   # vim /etc/asterisk/extensions.conf
+<code>; 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
+</code>
+   # asterisk -vvvvr
+<code>    -- 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'
+</code>
+   # vim /etc/asterisk/extensions.conf
+<code>; 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</code>
+   # asterisk -vvvvr
+<code>    -- 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'</code>
+   # vim /etc/asterisk/extensions.conf
+<code>; -----------------------------------------------------------------------------------------------------------------------
+; 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
+; ---------------------------------------------------------------------------------------------------------------------</code>
+   # asterisk -vvvvr
+<code>    -- 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'
+</code>
+   # vim /etc/asterisk/extensions.conf
+<code>; -----------------------------------------------------------------------------------------------------------------------
+; 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
+; -----------------------------------------------------------------------------------------------------------------------
+</code>
+   # asterisk -vvvvr
+<code>    -- 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'</code>
+   # vim /etc/asterisk/extensions.conf
+<code>; -----------------------------------------------------------------------------------------------------------------------
+; 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
+; -----------------------------------------------------------------------------------------------------------------------
+</code>
+   # asterisk -vvvvr
+<code>    -- 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'
+</code>
+<code>    -- 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'</code>
+   # vim /etc/asterisk/extensions.conf
+<code>; -----------------------------------------------------------------------------------------------------------------------
+; 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
+; -----------------------------------------------------------------------------------------------------------------------
+</code>
+   # asterisk -vvvvr
+<code>    -- 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'</code>
+<code>    -- 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'</code>
+   # vim /etc/asterisk/extensions.conf
+<code>; -----------------------------------------------------------------------------------------------------------------------
+; 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
+; -----------------------------------------------------------------------------------------------------------------------</code>
+<code>    -- 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'
+</code>
+<code>    -- 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'</code>
+   # vim /etc/asterisk/extensions.conf
+<code>; -----------------------------------------------------------------------------------------------------------------------
+; 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
+; -----------------------------------------------------------------------------------------------------------------------
+</code>
+   # asterisk -vvvvr
+<code>    -- 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></code>
+   # asterisk -vvvvr
+<code>    -- 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'
+</code>
+   # asterisk -vvvvr
+<code>    -- 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'</code>
+===== deutsche Sprachprompts =====
+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
+<file bash /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
+</file>
+   # chmod +x /usr/local/bin/asterisk-sound-convert
+   # asterisk-sound-convert
+===== Music on hold =====
+   # yum install madplay -y
+https://www.asteriskguru.com/tools/audio_conversion.php
+===== Anbindung an das VoIP-Netz der Telekom mit PJSIP =====
+==== Grundkonfiguration ====
+==== individuelle Rufnummer bei der abgehenden Wahl setzen ====
+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 **CLID**((**C**alling **L**ine **ID**entification)) 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. LOL
+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
+===== Voicemail =====
+   # 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
+<file bash /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
+</file>
+   # yum install lame -y
+===== Fax =====
+   # yum install asterisk-fax
+   # rpm -qil asterisk-fax
+<code>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</code>