Unterschiede

Hier werden die Unterschiede zwischen zwei Versionen angezeigt.

--- voip:c7:asterisk14:start [18.01.2017 19:04. ] – [Asterisk-Konsole] django
+++ voip:c7:asterisk14:start [18.11.2024 06:56. ] (aktuell) – Externe Bearbeitung 127.0.0.1
@@ Zeile 7: / Zeile 7: @@
 ===== 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 34: / 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 121: / 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 149: / Zeile 149: @@
 Build Host  : olive.ph.tucny.com
 Relocations : (not relocatable)
-URL         : http://www.asterisk.org/
+URL         : https://www.asterisk.org/
 Summary     : The Open Source PBX
 Description :
@@ Zeile 671: / Zeile 671: @@
 Build Host  : olive.ph.tucny.com
 Relocations : (not relocatable)
-URL         : http://www.asterisk.org/
+URL         : https://www.asterisk.org/
 Summary     : IAX2 channel driver for Asterisk
 Description :
@@ Zeile 681: / Zeile 681: @@
 /usr/share/asterisk/firmware/iax</code>
-=== asterisk-sip ===
+=== 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]]**. Wie installieren also auch hier das passende **RPM**.
+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.
-   # yum install asterisk-sip
-Was genau diese RPM-Paket in unserern Server mitgebracht hat zeigt uns wie immer der Aufruf des Befehls **rpm** mit der Option **-qil**.
+Wie installieren also auch hier das passende **RPM**.
-   # rpm -qil asterisk-sip
+   # yum install asterisk-pjsip
-<code>Name        : asterisk-sip
+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: Wed 18 Jan 2017 07:15:08 PM CET
+Install Date: Thu 19 Jan 2017 02:21:55 PM CET
 Group       : Applications/Internet
-Size        : 871899
+Size        : 1554633
 License     : GPLv2
 Signature   : RSA/SHA1, Thu 08 Dec 2016 11:58:06 PM CET, Key ID ccf40e16b9a46fa9
@@ Zeile 701: / Zeile 703: @@
 Build Host  : olive.ph.tucny.com
 Relocations : (not relocatable)
-URL         : http://www.asterisk.org/
+URL         : https://www.asterisk.org/
-Summary     : Legacy SIP channel driver for Asterisk
+Summary     : SIP channel based upon the PJSIP library
 Description :
-Legacy SIP channel driver for Asterisk
+SIP channel based upon the PJSIP library
-/etc/asterisk/sip.conf
+/etc/asterisk/pjproject.conf
-/etc/asterisk/sip_notify.conf
+/etc/asterisk/pjsip.conf
-/usr/lib64/asterisk/modules/chan_sip.so</code>
+/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 =====
@@ Zeile 1110: / Zeile 1163: @@
   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>