{"id":15726002,"url":"https://github.com/altanai/asteriskexamples","last_synced_at":"2026-02-07T02:33:38.872Z","repository":{"id":49997809,"uuid":"283940074","full_name":"altanai/asteriskexamples","owner":"altanai","description":null,"archived":false,"fork":false,"pushed_at":"2022-12-11T20:42:57.000Z","size":1535,"stargazers_count":10,"open_issues_count":4,"forks_count":3,"subscribers_count":3,"default_branch":"master","last_synced_at":"2025-07-26T23:44:59.318Z","etag":null,"topics":["asterisk","chan-pjsip","chan-sip","dialplans","pjsip"],"latest_commit_sha":null,"homepage":"https://telecom.altanai.com/category/session-initiation-prot-sip/sip-servers/asterisk/","language":"HTML","has_issues":true,"has_wiki":null,"has_pages":null,"mirror_url":null,"source_name":null,"license":null,"status":null,"scm":"git","pull_requests_enabled":true,"icon_url":"https://github.com/altanai.png","metadata":{"files":{"readme":"README.md","changelog":null,"contributing":null,"funding":null,"license":null,"code_of_conduct":null,"threat_model":null,"audit":null,"citation":null,"codeowners":null,"security":"security.md","support":null}},"created_at":"2020-07-31T04:16:08.000Z","updated_at":"2025-07-05T04:17:32.000Z","dependencies_parsed_at":"2023-01-27T06:45:45.678Z","dependency_job_id":null,"html_url":"https://github.com/altanai/asteriskexamples","commit_stats":null,"previous_names":[],"tags_count":0,"template":false,"template_full_name":null,"purl":"pkg:github/altanai/asteriskexamples","repository_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/altanai%2Fasteriskexamples","tags_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/altanai%2Fasteriskexamples/tags","releases_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/altanai%2Fasteriskexamples/releases","manifests_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/altanai%2Fasteriskexamples/manifests","owner_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/owners/altanai","download_url":"https://codeload.github.com/altanai/asteriskexamples/tar.gz/refs/heads/master","sbom_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories/altanai%2Fasteriskexamples/sbom","scorecard":null,"host":{"name":"GitHub","url":"https://github.com","kind":"github","repositories_count":286080680,"owners_count":29184978,"icon_url":"https://github.com/github.png","version":null,"created_at":"2022-05-30T11:31:42.601Z","updated_at":"2026-02-07T00:44:15.062Z","status":"online","status_checked_at":"2026-02-07T02:00:07.217Z","response_time":63,"last_error":null,"robots_txt_status":"success","robots_txt_updated_at":"2025-07-24T06:49:26.215Z","robots_txt_url":"https://github.com/robots.txt","online":true,"can_crawl_api":true,"host_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub","repositories_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repositories","repository_names_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/repository_names","owners_url":"https://repos.ecosyste.ms/api/v1/hosts/GitHub/owners"}},"keywords":["asterisk","chan-pjsip","chan-sip","dialplans","pjsip"],"created_at":"2024-10-03T22:25:21.999Z","updated_at":"2026-02-07T02:33:38.856Z","avatar_url":"https://github.com/altanai.png","language":"HTML","funding_links":[],"categories":[],"sub_categories":[],"readme":"# Asterisk \n\nAsterix open-source telephony Server can be used to build multitude of applications . \n\n## Installation and setup \n\ncheck the asterisk available version  from https://downloads.asterisk.org/pub/telephony/asterisk/\nuse wget to download \n\n```commandline\nwget https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-17.6.0.tar.gz\ntar -xvf asterisk-17.6.0.tar.gz\n```\n\n libpri library allows Asterisk to communicate with ISDN connections.\n \n DAHDI ( Digium Asterisk Hardware Device Interface) library allows Asterisk to communicate with analog and digital telephones and telephone lines / PSTN\n \n dahdi-linux \n ```commandline\nwget https://downloads.asterisk.org/pub/telephony/dahdi-linux/dahdi-linux-current.tar.gz\ntar -xvf dahdi-linux-current.tar.gz\n```\ndadhi-tools\n```commandline\nwget https://downloads.asterisk.org/pub/telephony/dahdi-tools/dahdi-tools-current.tar.gz \ntar -xvf dahdi-tools-current.tar.gz\n``` \n \nRemember you make to make and install DAHDI and libpri before building asterisk \n ```commandline\nmake\nsudo make install \nmake config \n```\n\n## Build asterisk  \n \nSIP stack ina Asterisk  \n- chan_sip SIP channel driver ( old)\n- chan_pjsip\n\nBuild with later using \n```commandline\ncd asterisk-17.6.0\n./configure --with-pjproject-bundled --with-jansson-bundled\n```\noutput\n```\nconfig.status: creating makeopts\nconfig.status: creating autoconfig.h\nconfigure: Menuselect build configuration successfully completed\n\n               .$$$$$$$$$$$$$$$=..      \n            .$7$7..          .7$$7:.    \n          .$$:.                 ,$7.7   \n        .$7.     7$$$$           .$$77  \n     ..$$.       $$$$$            .$$$7 \n    ..7$   .?.   $$$$$   .?.       7$$$.\n   $.$.   .$$$7. $$$$7 .7$$$.      .$$$.\n .777.   .$$$$$$77$$$77$$$$$7.      $$$,\n $$$~      .7$$$$$$$$$$$$$7.       .$$$.\n.$$7          .7$$$$$$$7:          ?$$$.\n$$$          ?7$$$$$$$$$$I        .$$$7 \n$$$       .7$$$$$$$$$$$$$$$$      :$$$. \n$$$       $$$$$$7$$$$$$$$$$$$    .$$$.  \n$$$        $$$   7$$$7  .$$$    .$$$.   \n$$$$             $$$$7         .$$$.    \n7$$$7            7$$$$        7$$$      \n $$$$$                        $$$       \n  $$$$7.                       $$  (TM)     \n   $$$$$$$.           .7$$$$$$  $$      \n     $$$$$$$$$$$$7$$$$$$$$$.$$$$$$      \n       $$$$$$$$$$$$$$$$.                \n\nconfigure: Package configured for: \nconfigure: OS type  : linux-gnu\nconfigure: Host CPU : x86_64\nconfigure: build-cpu:vendor:os: x86_64 : pc : linux-gnu :\nconfigure: host-cpu:vendor:os: x86_64 : pc : linux-gnu :\n```\n  \nUSe menu select option on Asterisk to install packages pick and choose application and other options such as music on hold , extra sound packages , PBX modules etc\n```commandline\nmake menuselect\n```  \noutput \n```commandline\n **************************************************\n     Asterisk Module and Build Option Selection\n **************************************************\n\n          Press 'h' for help.\n\n      Add-ons (See README-addons.txt)\n      Applications\n      Bridging Modules\n      Call Detail Recording\n      Channel Event Logging\n      Channel Drivers\n      Codec Translators\n      Format Interpreters\n      Dialplan Functions\n      PBX Modules\n      Resource Modules\n      Test Modules\n      Compiler Flags\n      Utilities\n      AGI Samples\n      Core Sound Packages\n      Music On Hold File Packages\n ---\u003e Extras Sound Packages\n```\n\nCompile asterisk on the system \n```commandline\nmake\n```\noutput \n```commandline\nBuilding Documentation For: third-party channels pbx apps codecs formats cdr cel bridges funcs tests main res addons \n +--------- Asterisk Build Complete ---------+\n + Asterisk has successfully been built, and +\n + can be installed by running:              +\n +                                           +\n +                make install               +\n +-------------------------------------------+\n\n```\n\nInstall \n```commandline\nsudo make install\n```\noutput\n```commandline\ndone\n +---- Asterisk Installation Complete -------+\n +                                           +\n +    YOU MUST READ THE SECURITY DOCUMENT    +\n +                                           +\n + Asterisk has successfully been installed. +\n + If you would like to install the sample   +\n + configuration files (overwriting any      +\n + existing config files), run:              +\n +                                           +\n + For generic reference documentation:      +\n +    make samples                           +\n +                                           +\n + For a sample basic PBX:                   +\n +    make basic-pbx                         +\n +                                           +\n +                                           +\n +-----------------  or ---------------------+\n\n```\n\nInstall samples \n```commandline\nsudo make samples \n```\n\nRun Asterisk\n```commandline\n➜  asterisk-17.6.0 sudo asterisk -c \nAsterisk 17.6.0, Copyright (C) 1999 - 2018, Digium, Inc. and others.\nCreated by Mark Spencer \u003cmarkster@digium.com\u003e\n...\nAsterisk Ready.\n*CLI\u003e \n```\nports uses \n````commandline\n\u003e ps -ef | grep asterisk\n\nasterisk  28922 altanai    5u  IPv4 3449570      0t0  TCP localhost:38108-\u003elocalhost:postgresql (CLOSE_WAIT)\nasterisk  28922 altanai    9u  IPv4 3451466      0t0  UDP *:40388 \nasterisk  28922 altanai   10u  IPv6 3451467      0t0  UDP *:52092 \nasterisk  28922 altanai   12u  IPv4 3449574      0t0  UDP *:2727 \nasterisk  28922 altanai   14u  IPv4 3449575      0t0  UDP *:iax \nasterisk  28922 altanai   15u  IPv4 3451995      0t0  TCP *:cisco-sccp (LISTEN)\nasterisk  28922 altanai   18u  IPv4 3449577      0t0  UDP *:5000 \nasterisk  28922 altanai   19u  IPv4 3449578      0t0  UDP *:4520 \n\n````\n\nYou can use safe_asterisk to enable auto restart after crash thus minimizing downtime. It also creates core dump file .\n\nOne can continue on the sandbox environment to run the modules I describe in this repo.\n\nshow version\n```commandline\n*CLI\u003e core show version \nAsterisk 17.6.0 built by root @ altanai-Inspiron-15-5578 on a x86_64 running Linux on 2020-07-30 15:33:43 UTC\n```\n\nAsterisk is an open source framework for building communications applications.\n\n**Default installation paths**\nPath Description\n/etc/asterisk Configuration files\n/usr/sbin Location of binary executable\n/var/log/asterisk message(error) logs and CDR\n/usr/lib/asterisk/modules Component module libraries\n\n**Default ports**\n\nProtocol Port number Transport\nSIP 5060/5061 TCP/UDP\nIAX2 4569 UDP\nMGCP 2727 UDP\nSCCP 2000 TCP\nRTP 10,00 – 20,000 UDP\nManager 5038 TCP\nH323 1720 TCP\nDundi 4520 UDP\nUnistim 5000 UDP\n\n## configuration \n\nListen address for SIP in sip.conf , Listen on\n\nspecific IPv4 address.      Example: bindaddr=192.0.2.1\nspecific IPv6 address.      Example: bindaddr=2001:db8::1\nIPv4 wildcard.              Example: bindaddr=0.0.0.0\nIPv4 and IPv6 wildcards.    Example: bindaddr=::\n\n## Modules \n\nasterisk/modules support teh modular structure of Asterisk\n\n## Resources \n\nCustom applications such as res_musiconhold.so\n\n## Codecs and Formats \n\nconvert media on disk and channel \n\n## Cli commands \n\nInterface with asterisk and issue commands\n\nOther interface for asterisk control are AMI ( Asterisk AManger Interface) and AGI ( Asterisk Gateway Interface ) which operate on APIs like PHP , C++ , Java , Perl\n\nReload SIP\n```commandline\n*CLI\u003e pjsip reload\n[Jul 31 09:54:58] NOTICE[22399]: sorcery.c:1345 sorcery_object_load: Type 'system' is not reloadable, maintaining previous values\nModule 'res_pjsip.so' reloaded successfully.\nModule 'res_pjsip_authenticator_digest.so' reloaded successfully.\nModule 'res_pjsip_endpoint_identifier_ip.so' reloaded successfully.\nModule 'res_pjsip_mwi.so' reloaded successfully.\nModule 'res_pjsip_notify.so' reloaded successfully.\nModule 'res_pjsip_outbound_publish.so' reloaded successfully.\nModule 'res_pjsip_publish_asterisk.so' reloaded successfully.\nModule 'res_pjsip_outbound_registration.so' reloaded successfully.\n```\nSettings show \n```commandline\n*CLI\u003e core show settings\n\nPBX Core settings\n-----------------\n  Version:                     17.6.0\n  Build Options:               BUILD_NATIVE, OPTIONAL_API\n  Maximum calls:               Not set\n  Maximum open file handles:   1024\n  Root console verbosity:      0\n  Current console verbosity:   0\n  Debug level:                 0\n  Trace level:                 0\n  Maximum load average:        0.000000\n  Minimum free memory:         0 MB\n  Startup time:                10:33:28\n  Last reload time:            10:33:28\n  System:                      Linux/4.15.0-62-generic built by root on x86_64 2020-07-30 15:33:43 UTC\n  System name:                 \n  Entity ID:                   7c:67:a2:eb:ff:a5\n  PBX UUID:                    fc3a2e05-e800-47e7-aa27-36ad924f85e0\n  Default language:            en\n  Language prefix:             Enabled\n  User name and group:         /\n  Executable includes:         Disabled\n  Transcode via SLIN:          Enabled\n  Transmit silence during rec: Disabled\n  Generic PLC:                 Enabled\n  Generic PLC on equal codecs: Disabled\n  Hide Msg Chan AMI events:    Disabled\n  Min DTMF duration::          80\n  Cache media frames:          Enabled\n  RTP use dynamic payloads:    1\n  RTP dynamic payload types:   35-63,96-127\n\n* Subsystems\n  -------------\n  Manager (AMI):               Disabled\n  Web Manager (AMI/HTTP):      Disabled\n  Call data records:           Enabled\n  Realtime Architecture (ARA): Disabled\n\n* Directories\n  -------------\n  Configuration file:          /etc/asterisk/asterisk.conf\n  Configuration directory:     /etc/asterisk\n  Module directory:            /usr/lib/asterisk/modules\n  Spool directory:             /var/spool/asterisk\n  Log directory:               /var/log/asterisk\n  Run/Sockets directory:       /var/run/asterisk\n  PID file:                    /var/run/asterisk/asterisk.pid\n  VarLib directory:            /var/lib/asterisk\n  Data directory:              /var/lib/asterisk\n  ASTDB:                       /var/lib/asterisk/astdb\n  IAX2 Keys directory:         /var/lib/asterisk/keys\n  AGI Scripts directory:       /var/lib/asterisk/agi-bin\n\n```\n\nchannel types \n```commandline\n*CLI\u003e core show channeltypes\nType             Description                              Devicestate   Presencestate Indications   Transfer     \n-------------    -------------                            ------------- ------------- ------------- -------------\nRecorder         Bridge Media Recording Channel Driver    no            no            yes           no           \nAnnouncer        Bridge Media Announcing Channel Driver   no            no            yes           no           \nPhone            Standard Linux Telephony API Driver      no            no            yes           no           \nConsole          OSS Console Channel Driver               no            no            yes           no           \nUSTM             UNISTIM Channel Driver                   no            no            yes           no           \nCBAnn            Conference Bridge Announcing Channel     no            no            yes           no           \nCBRec            Conference Bridge Recording Channel      no            no            no            no           \nPJSIP            PJSIP Channel Driver                     yes           no            yes           yes          \nUnicastRTP       Unicast RTP Media Channel Driver         no            no            no            no           \nMulticastRTP     Multicast RTP Paging Channel Driver      no            no            no            no           \nSkinny           Skinny Client Control Protocol (Skinny)  yes           no            yes           no           \nIAX2             Inter Asterisk eXchange Driver (Ver 2)   yes           no            yes           yes          \nMGCP             Media Gateway Control Protocol (MGCP)    yes           no            yes           no           \nLocal            Local Proxy Channel Driver               yes           no            yes           no           \nSurrogate        Surrogate channel used to pull channel f no            no            no            no           \n----------\n15 channel drivers registered.\n\n```\n\nSet verbosity level ( 0-4)\n```commandline\n*CLI\u003e core set verbose 1\nConsole verbose was OFF and is now 1.\n```\n\n## Applications \nCustom application that can be built - Call Queue monitoring \n\nAdd a dialplan in extensions \n```\n[internal_users]\nexten =\u003e 6000,1,Answer()\nexten =\u003e 6000,2,Wait(1)\nexten =\u003e 6000,2,Wait(1)\nexten =\u003e 6000,3,Playback(hello-world)\nexten =\u003e 6000,4,Hangup()\n```\nReload and see the dialplan \n```commandline\naltanai-Inspiron-15-5578*CLI\u003e dialplan reload\nDialplan reloaded.\naltanai-Inspiron-15-5578*CLI\u003e dialplan show \n...\n[ Context 'internal_users' created by 'pbx_config' ]\n  '6000' =\u003e         1. Answer()                                   [extensions.conf:866]\n                    2. Wait(1)                                    [extensions.conf:867]\n                    3. Playback(hello-world)                      [extensions.conf:868]\n                    4. Hangup()                                   [extensions.conf:869]\n```\n\nDuring Call \n```commandline\n*CLI\u003e   == Setting global variable 'SIPDOMAIN' to '192.168.1.114'\n  == Spawn extension (internal_users, 6000, 5) exited non-zero on 'PJSIP/7000-00000000'\n```\n\nDial two Calls \n\nextensions.conf\n```\nexten =\u003e 6001,1,Dial(SIP/phone-2,20)\nexten =\u003e 6001,2,Verbose(\"---------------- Forward to phone2  -------\")\n```\nreload dialplan and validate that its taken effect \n```commandline\n*CLI\u003e dialplan show 6001@internal_users\n[ Context 'internal_users' created by 'pbx_config' ]\n  '6001' =\u003e         1. Dial(SIP/phone-2,20)                       [extensions.conf:872]\n                    2. Verbose(\"---------------- Forward to phone2  -------\") [extensions.conf:873]\n\n-= 1 extension (2 priorities) in 1 context. =-\n```\nDial 6001 and see channels while call is active\n```commandline\ncore show channels\nChannel              Location             State   Application(Data)             \n0 active channels\n0 active calls\n1 call processed\n```\n\nNote that the sounds for asterisk playback , background and various other applications are loaded in \n/var/lib/asterisk/sounds/en/\n\nMore core and extra sounds can be downloaded from https://www.asterisksounds.org/en/download in various other languages like german apnish italian etc\nsuch as \n```commandline\nwget https://www.asterisksounds.org/sites/asterisksounds.org/files/sounds/en/download/asterisk-sounds-extra-en-2.9.15.zip\n```\n\n## debugging \n\n**Issue1** Cannot find sip commands \n```commandline\naltanai-Inspiron-15-5578*CLI\u003e sip show peers\nNo such command 'sip show peers' (type 'core show help sip show' for other possible commands)\n```\n\\\n**solution** First check is module is installed \n```commandline\nmodule show like chan_sip.so \n```\nWhen module is installed \n```shell script\naltanai-Inspiron-15-5578*CLI\u003e module show like chan_sip.so\nModule                         Description                              Use Count  Status      Support Level\nchan_sip.so                    Session Initiation Protocol (SIP)        0         \n1 modules loaded\n```\n\nwhen modules is not installed \n```shell script\naltanai-Inspiron-15-5578*CLI\u003e module show like chan_sip.so\nModule                         Description                              Use Count  Status      Support Level\n0 modules loaded\n```\n\nIf module is not loaded Make sure Asterisk is configured to load the module via modules.conf\nThere may be a case that res_pjsip is loaded inplace of chan_sip\n```\n; Do not load chan_sip by default, it may conflict with res_pjsip.\nnoload =\u003e chan_sip.so\n```\nIn that case chan_pjsip would be loaded \n```commandline\naltanai-Inspiron-15-5578*CLI\u003e module show like chan_pjsip\nModule                         Description                              Use Count  Status      Support Level\nchan_pjsip.so                  PJSIP Channel Driver                     0          Running              core\n1 modules loaded\n```\nOn forcing loading of chan_sip module\n```commandline\naltanai-Inspiron-15-5578*CLI\u003e module load chan_sip\nLoaded chan_sip\nSIP channel loading...\n[Jul 31 13:25:30] WARNING[815]: chan_sip.c:35477 deprecation_notice: chan_sip has no official maintainer and is deprecated.  Migration to\n[Jul 31 13:25:30] WARNING[815]: chan_sip.c:35478 deprecation_notice: chan_pjsip is recommended.  See guides at the Asterisk Wiki:\n[Jul 31 13:25:30] WARNING[815]: chan_sip.c:35479 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip\n[Jul 31 13:25:30] WARNING[815]: chan_sip.c:35480 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip\n\n```\n\n**Issue3** Unable to authenticate \n```commandline\n*CLI\u003e m[Aug  1 10:07:20] NOTICE[11076]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '\u003csip:phone-1@127.0.0.1\u003e' failed for '127.0.0.1:56039' (callid: OgfL6JO~OO) - No matching endpoint found\n[Aug  1 10:07:25] NOTICE[11076]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '\u003csip:phone-1@127.0.0.1\u003e' failed for '127.0.0.1:44672' (callid: c62tK1JzlDnDS9VFDGMQfg..) - No matching endpoint found\n[Aug  1 10:07:25] NOTICE[11076]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '\u003csip:phone-1@127.0.0.1\u003e' failed for '127.0.0.1:44672' (callid: c62tK1JzlDnDS9VFDGMQfg..) - No matching endpoint found\n[Aug  1 10:07:25] NOTICE[11076]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'INVITE' from '\u003csip:phone-1@127.0.0.1\u003e' failed for '127.0.0.1:44672' (callid: c62tK1JzlDnDS9VFDGMQfg..) - Failed to authenticate\n```\n\\\n**solution** Define the extension context internal_users\nextensions.conf\n```\n[internal_users]\nexten =\u003e 6000,1,Answer()\nexten =\u003e 6000,2,Wait(1)\nexten =\u003e 6000,3,Playback(hello-world)\nexten =\u003e 6000,4,Hangup()\n\n```\n\nand edit sip.conf or pjsip.conf and reload \n\nsip.conf\n```\n[7000]\n    type=friend\n    host=dynamic\n    context=internal_users\n    secret=1234\n```\n\npjsip.conf\n```\n[7000]\n type=endpoint\n context=internal_users\n disallow=all\n allow=ulaw\n transport=transport-udp\n auth=7000\n aors=7000\n\n[7000]\n type=auth\n auth_type=userpass\n password=1234\n username=7000\n\n[7000]\n type=aor\n max_contacts=1\n\n```\nreload \n```commandline\nsip reload \npjsip reload\n```\n\n**Issue3** Unable to play hello world via playback\n\\\n**solution** check if application is registered \n```commandline\ncore show application playback\n```\nIf registered \n```commandline\n  -= Info about application 'Playback' =- \n\n[Synopsis]\nPlay a file. \n\n[Description]\nPlays back given filenames (do not put extension of wav/alaw etc). The playback\ncommand answer the channel if no options are specified. If the file is\nnon-existant it will fail\nThis application sets the following channel variable upon completion:\n${PLAYBACKSTATUS}: The status of the playback attempt as a text string.\n    SUCCESS\n    FAILED\nSee Also: Background (application) -- for playing sound files that are\ninterruptible\nWaitExten (application) -- wait for digits from caller, optionally play music\non hold\n\n[Syntax]\nPlayback(filename[\u0026filename2[\u0026...]][,options])\n\n[Arguments]\noptions\n    Comma separated list of options\n    skip: Do not play if not answered\n\n    noanswer: Playback without answering, otherwise the channel will be\n    answered before the sound is played.\n    NOTE: Not all channel types support playing messages while still on hook.\n\n\n[See Also]\nBackground(), WaitExten(), ControlPlayback(), stream file, control stream file,\n```\n\nIf not registered \n```commandline\naltanai-Inspiron-15-5578*CLI\u003e core show application playback\nYour application(s) is (are) not registered\n```\n\n\n**Issue 4** Public / private address conflicts \n![Sceen1](screenshots/screenshot1.png)\n\\\n**solution** Change bidn addr from 0.0.0.0 to private IP and reload \n![Sceen1](screenshots/screenshot2.png)\n\n**Issue5** While working on IVR , WaitExtern , Background and DTMF applications you may find the prompt and press sounds missing \n```commandline\n[Aug  2 08:55:44] WARNING[25971][C-0000002d]: file.c:1262 ast_streamfile: Unable to open are-you-still-there (format (ulaw)): No such file or directory\n[Aug  2 08:55:44] WARNING[25971][C-0000002d]: app_playback.c:497 playback_exec: Playback failed on PJSIP/7000-0000002b for are-you-still-there\n```\nor \n```commandline\n[Aug  2 08:54:52] WARNING[25971][C-0000002d]: file.c:789 ast_openstream_full: File press-1 does not exist in any format\n[Aug  2 08:54:52] WARNING[25971][C-0000002d]: file.c:1262 ast_streamfile: Unable to open press-1 (format (ulaw)): No such file or directory\n[Aug  2 08:54:52] WARNING[25971][C-0000002d]: pbx_builtins.c:1175 pbx_builtin_background: ast_streamfile failed on PJSIP/7000-0000002b for press-1\u0026or\u0026press-2\n```\n**solution** Download sound files or make your own prompts \n","project_url":"https://awesome.ecosyste.ms/api/v1/projects/github.com%2Faltanai%2Fasteriskexamples","html_url":"https://awesome.ecosyste.ms/projects/github.com%2Faltanai%2Fasteriskexamples","lists_url":"https://awesome.ecosyste.ms/api/v1/projects/github.com%2Faltanai%2Fasteriskexamples/lists"}