Hi, I 'm trying to play sound file with playback() function.
But sound is not coming.
Any suggetion would be helpful.
Thanks regard.
here is my configurations.
/etc/asterisk/sip.conf
; SIP Configuration for Asterisk
[general]
context => public ; Default context for incoming calls. Defaults to 'default'
allowguest => no ; Allow or reject guest calls (default is yes)
allowoverlap => no ; Disable overlap dialing support. (Default is yes)
tcpenable => yes ; Enable server for incoming TCP connections (default is no)
tcpbindaddr => 0.0.0.0:15060 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
udpbindaddr => 0.0.0.0:15060 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
transport => udp ; Set the default transports. The order determines the primary default transport.
localnet => XXXXXXXXXXXXXXXX
externip => XXXXXXXXXXXXXXXX
srvlookup => yes ; Enable DNS SRV lookups on outbound calls
language => ja ; sDefault language setting for all users/peers
rtcachefriends => yes ; realtime database settings
rtautoclear => yes ; realtime database settings
/etc/asterisk/extensions.conf
; the Asterisk dial plan
[general]
static => yes
writeprotect => no
clearglobalvars => no
[globals]
CONSOLE => Console/dsp
IAXINFO => guest
TRUNK => DAHDI/G2
TRUNKMSD => 1
#include additions/extensions.conf
/etc/asterisk/additions/extensions.conf
[ctx_marshal-i]
exten => _marshalai_.,1,noop(${CALLERID(dnid)} -> ${EXTEN})
; define
;same => n,set(__MAXCALL=201)
same => n,set(__MAXCALL=1)
; gather_info
same => n,agi(/var/lib/asterisk/agi-bin/marshal-i/gather_info.php,${EXTEN})
same => n,noop(CENTER=${CENTER})
same => n,noop(ACCOUNT=${ACCOUNT})
same => n,set(GROUP()=NUMBER_OF_CALLS)
; check to reach max number of calls
same => n,gotoif($[${GROUP_COUNT(NUMBER_OF_CALLS)} < ${MAXCALL}]?lbl_continue:lbl_play_busy)
; count up number of calls
same => n(lbl_continue),noop
same => n,noop(current number of calls=${GROUP_COUNT(NUMBER_OF_CALLS)})
; make call
same => n,dial(SIP/${EXTEN}/${EXTEN})
same => n,goto(lbl_hang)
; play congestion
same => n(lbl_play_busy),gosub(ctx_common,play_congestion,1)
same => n,goto(lbl_hang)
; hang
same => n(lbl_hang),hangup
[ctx_common]
exten => play_congestion,1,noop
same => n,answer
same => n,playback(all-circuits-busy-now)
same => n,return
This is asterisk CLI log when I make a call. That looks be fine. There is no error.
asterisk1-dev*CLI>
== Using SIP RTP CoS mark 5
-- Executing [marshalai_1_2@ctx_marshal-i:1] NoOp("SIP/marshalai_1_1-00000001", "marshalai_1_2 -> marshalai_1_2") in new stack
-- Executing [marshalai_1_2@ctx_marshal-i:2] Set("SIP/marshalai_1_1-00000001", "__MAXCALL=1") in new stack
-- Executing [marshalai_1_2@ctx_marshal-i:3] AGI("SIP/marshalai_1_1-00000001", "/var/lib/asterisk/agi-bin/marshal-i/gather_info.php,marshalai_1_2") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/marshal-i/gather_info.php
-- <SIP/marshalai_1_1-00000001>AGI Script /var/lib/asterisk/agi-bin/marshal-i/gather_info.php completed, returning 0
-- Executing [marshalai_1_2@ctx_marshal-i:4] NoOp("SIP/marshalai_1_1-00000001", "CENTER=1") in new stack
-- Executing [marshalai_1_2@ctx_marshal-i:5] NoOp("SIP/marshalai_1_1-00000001", "ACCOUNT=2") in new stack
-- Executing [marshalai_1_2@ctx_marshal-i:6] Set("SIP/marshalai_1_1-00000001", "GROUP()=NUMBER_OF_CALLS") in new stack
-- Executing [marshalai_1_2@ctx_marshal-i:7] GotoIf("SIP/marshalai_1_1-00000001", "0?lbl_continue:lbl_play_busy") in new stack
-- Goto (ctx_marshal-i,marshalai_1_2,12)
-- Executing [marshalai_1_2@ctx_marshal-i:12] Gosub("SIP/marshalai_1_1-00000001", "ctx_common,play_congestion,1") in new stack
-- Executing [play_congestion@ctx_common:1] NoOp("SIP/marshalai_1_1-00000001", "") in new stack
-- Executing [play_congestion@ctx_common:2] Answer("SIP/marshalai_1_1-00000001", "") in new stack
> 0x7f70c8003780 -- Probation passed - setting RTP source address to 202.215.229.97:62547
-- Executing [play_congestion@ctx_common:3] Playback("SIP/marshalai_1_1-00000001", "all-circuits-busy-now") in new stack
-- <SIP/marshalai_1_1-00000001> Playing 'all-circuits-busy-now.ulaw' (language 'ja')
> 0x7f70c8003780 -- Probation passed - setting RTP source address to 202.215.229.97:62547
-- Executing [play_congestion@ctx_common:4] Return("SIP/marshalai_1_1-00000001", "") in new stack
-- Executing [marshalai_1_2@ctx_marshal-i:13] Goto("SIP/marshalai_1_1-00000001", "lbl_hang") in new stack
-- Goto (ctx_marshal-i,marshalai_1_2,14)
-- Executing [marshalai_1_2@ctx_marshal-i:14] Hangup("SIP/marshalai_1_1-00000001", "") in new stack
== Spawn extension (ctx_marshal-i, marshalai_1_2, 14) exited non-zero on 'SIP/marshalai_1_1-00000001'
I also check loaded modules.
asterisk1-dev*CLI> module show
Module Description Use Count Status Support Level
app_alarmreceiver.so Alarm Receiver for Asterisk 0 Not Running extended
app_authenticate.so Authentication Application 0 Running core
app_bridgewait.so Place the channel into a holding bridge 0 Running core
app_celgenuserevent.so Generate an User-Defined CEL event 0 Running core
app_chanisavail.so Check channel availability 0 Running extended
app_channelredirect.so Redirects a given channel to a dialplan 0 Running core
app_chanspy.so Listen to the audio of an active channel 0 Running core
app_controlplayback.so Control Playback Application 0 Running core
app_db.so Database Access Functions 0 Running core
app_dial.so Dialing Application 0 Running core
app_dictate.so Virtual Dictation Machine 0 Running extended
app_directed_pickup.so Directed Call Pickup Application 0 Running core
app_directory.so Extension Directory 0 Running core
app_disa.so DISA (Direct Inward System Access) Appli 0 Running core
app_dumpchan.so Dump Info About The Calling Channel 0 Running core
app_echo.so Simple Echo Application 0 Running core
app_exec.so Executes dialplan applications 0 Running core
app_externalivr.so External IVR Interface Application 0 Running extended
app_getcpeid.so Get ADSI CPE ID 0 Running extended
app_ices.so Encode and Stream via icecast and ices 0 Running extended
app_image.so Image Transmission Application 0 Running extended
app_macro.so Extension Macros 0 Running core
app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 Running core
app_mixmonitor.so Mixed Audio Monitoring Application 0 Running core
app_morsecode.so Morse code 0 Running extended
app_mp3.so Silly MP3 Application 0 Running extended
app_nbscat.so Silly NBS Stream Application 0 Running extended
app_originate.so Originate call 0 Running core
app_page.so Page Multiple Phones 0 Running core
app_playback.so Sound File Playback Application 0 Running core <------ it's running
app_playtones.so Playtones Application 0 Running core
app_privacy.so Require phone number to be entered, if n 0 Running core
app_read.so Read Variable Application 0 Running core
app_readexten.so Read and evaluate extension validity 0 Running core
app_record.so Trivial Record Application 0 Running core
app_sayunixtime.so Say time 0 Running core
app_senddtmf.so Send DTMF digits Application 0 Running core
app_sendtext.so Send Text Applications 0 Running core
app_sms.so SMS/PSTN handler 0 Running extended
app_softhangup.so Hangs up the requested channel 0 Running core
app_speech_utils.so Dialplan Speech Applications 0 Running core
app_stack.so Dialplan subroutines (Gosub, Return, etc 0 Running core
app_stasis.so Stasis dialplan application 0 Running core
app_system.so Generic System() application 0 Running core
app_talkdetect.so Playback with Talk Detection 0 Running extended
app_test.so Interface Test Application 0 Running extended
app_transfer.so Transfers a caller to another extension 0 Running core
app_url.so Send URL Applications 0 Running extended
app_userevent.so Custom User Event Application 0 Running core
app_verbose.so Send verbose output 0 Running core
app_waitforring.so Waits until first ring after time 0 Running extended
app_waitforsilence.so Wait For Silence 0 Running extended
app_waituntil.so Wait until specified time 0 Running core
app_while.so While Loops and Conditional Execution 0 Running core
app_zapateller.so Block Telemarketers with Special Informa 0 Running extended
bridge_builtin_features.so Built in bridging features 1 Running core
bridge_builtin_interval_features.so Built in bridging interval features 0 Running core
bridge_holding.so Holding bridge module 0 Running core
bridge_native_rtp.so Native RTP bridging module 0 Running core
bridge_simple.so Simple two channel bridging module 0 Running core
bridge_softmix.so Multi-party software based channel mixin 0 Running core
chan_bridge_media.so Bridge Media Channel Driver 0 Running core
chan_rtp.so RTP Media Channel 0 Running core
chan_sip.so Session Initiation Protocol (SIP) 0 Running core
codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 Running core
codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 Running core
codec_alaw.so A-law Coder/Decoder 0 Running core
codec_g722.so ITU G.722-64kbps G722 Transcoder 0 Running core
codec_g726.so ITU G.726-32kbps G726 Transcoder 0 Running core
codec_gsm.so GSM Coder/Decoder 0 Running core
codec_ilbc.so iLBC Coder/Decoder 0 Running core
codec_lpc10.so LPC10 2.4kbps Coder/Decoder 0 Running core
codec_resample.so SLIN Resampling Codec 0 Running core
codec_ulaw.so mu-Law Coder/Decoder 0 Running core
format_g719.so ITU G.719 0 Running core
format_g723.so G.723.1 Simple Timestamp File Format 0 Running core
format_g726.so Raw G.726 (16/24/32/40kbps) data 0 Running core
format_g729.so Raw G.729 data 0 Running core
format_gsm.so Raw GSM data 0 Running core
format_h263.so Raw H.263 data 0 Running core
format_h264.so Raw H.264 data 0 Running core
format_ilbc.so Raw iLBC data 0 Running core
format_jpeg.so jpeg (joint picture experts group) image 0 Running extended
format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 Running core
format_siren14.so ITU G.722.1 Annex C (Siren14, licensed f 0 Running core
format_siren7.so ITU G.722.1 (Siren7, licensed from Polyc 0 Running core
format_sln.so Raw Signed Linear Audio support (SLN) 8k 0 Running core
format_vox.so Dialogic VOX (ADPCM) File Format 0 Running extended
format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 Running core
format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 Running core
func_aes.so AES dialplan functions 0 Running core
func_audiohookinherit.so Audiohook inheritance placeholder functi 0 Running deprecated
func_base64.so base64 encode/decode dialplan functions 0 Running core
func_blacklist.so Look up Caller*ID name/number from black 0 Running core
func_callcompletion.so Call Control Configuration Function 0 Running core
func_callerid.so Party ID related dialplan functions (Cal 0 Running core
func_cdr.so Call Detail Record (CDR) dialplan functi 0 Running core
func_channel.so Channel information dialplan functions 0 Running core
func_config.so Asterisk configuration file variable acc 0 Running core
func_curl.so Load external URL 0 Running core
func_cut.so Cut out information from a string 0 Running core
func_db.so Database (astdb) related dialplan functi 0 Running core
func_devstate.so Gets or sets a device state in the dialp 0 Running core
func_dialgroup.so Dialgroup dialplan function 0 Running core
func_dialplan.so Dialplan Context/Extension/Priority Chec 0 Running core
func_enum.so ENUM related dialplan functions 0 Running core
func_env.so Environment/filesystem dialplan function 0 Running core
func_extstate.so Gets an extension's state in the dialpla 0 Running core
func_frame_trace.so Frame Trace for internal ast_frame debug 0 Running extended
func_global.so Variable dialplan functions 0 Running core
func_groupcount.so Channel group dialplan functions 0 Running core
func_hangupcause.so HANGUPCAUSE related functions and applic 0 Running core
func_holdintercept.so Hold interception dialplan function 0 Running core
func_iconv.so Charset conversions 0 Running core
func_jitterbuffer.so Jitter buffer for read side of channel. 0 Running core
func_lock.so Dialplan mutexes 0 Running core
func_logic.so Logical dialplan functions 0 Running core
func_math.so Mathematical dialplan function 0 Running core
func_md5.so MD5 digest dialplan functions 0 Running core
func_module.so Checks if Asterisk module is loaded in m 0 Running core
func_periodic_hook.so Periodic dialplan hooks. 0 Running core
func_pitchshift.so Audio Effects Dialplan Functions 0 Running extended
func_presencestate.so Gets or sets a presence state in the dia 0 Running core
func_rand.so Random number dialplan function 0 Running core
func_realtime.so Read/Write/Store/Destroy values from a R 0 Running core
func_sha1.so SHA-1 computation dialplan function 0 Running core
func_shell.so Collects the output generated by a comma 0 Running core
func_sorcery.so Get a field from a sorcery object 0 Running core
func_sprintf.so SPRINTF dialplan function 0 Running core
func_srv.so SRV related dialplan functions 0 Running core
func_strings.so String handling dialplan functions 0 Running core
func_sysinfo.so System information related functions 0 Running core
func_talkdetect.so Talk detection dialplan function 0 Running core
func_timeout.so Channel timeout dialplan functions 0 Running core
func_uri.so URI encode/decode dialplan functions 0 Running core
func_version.so Get Asterisk Version/Build Info 0 Running core
func_vmcount.so Indicator for whether a voice mailbox ha 0 Running core
func_volume.so Technology independent volume control 0 Running core
pbx_config.so Text Extension Configuration 0 Running core
pbx_loopback.so Loopback Switch 0 Running core
pbx_realtime.so Realtime Switch 0 Running extended
pbx_spool.so Outgoing Spool Support 0 Running core
res_adsi.so ADSI Resource 0 Running core
res_ael_share.so share-able code for AEL 0 Running extended
res_agi.so Asterisk Gateway Interface (AGI) 1 Running core
res_clioriginate.so Call origination and redirection from th 0 Running core
res_config_curl.so Realtime Curl configuration 0 Running core
res_config_odbc.so Realtime ODBC configuration 0 Running core
res_convert.so File format conversion CLI command 0 Running core
res_crypto.so Cryptographic Digital Signatures 1 Running core
res_curl.so cURL Resource Module 0 Running core
res_format_attr_celt.so CELT Format Attribute Module 1 Running core
res_format_attr_h263.so H.263 Format Attribute Module 1 Running core
res_format_attr_h264.so H.264 Format Attribute Module 1 Running core
res_format_attr_opus.so Opus Format Attribute Module 1 Running core
res_format_attr_silk.so SILK Format Attribute Module 1 Running core
res_format_attr_siren14.so Siren14 Format Attribute Module 1 Running core
res_format_attr_siren7.so Siren7 Format Attribute Module 1 Running core
res_format_attr_vp8.so VP8 Format Attribute Module 1 Running core
res_http_websocket.so HTTP WebSocket Support 1 Running extended
res_limit.so Resource limits 0 Running core
res_manager_devicestate.so Manager Device State Topic Forwarder 0 Running core
res_manager_presencestate.so Manager Presence State Topic Forwarder 0 Running core
res_monitor.so Call Monitoring Resource 0 Running core
res_mutestream.so Mute audio stream resources 0 Running core
res_odbc.so ODBC resource 0 Running core
res_odbc_transaction.so ODBC transaction resource 0 Running core
res_realtime.so Realtime Data Lookup/Rewrite 0 Running core
res_rtp_asterisk.so Asterisk RTP Stack 0 Running core
res_rtp_multicast.so Multicast RTP Engine 0 Running core
res_security_log.so Security Event Logging 0 Running core
res_sorcery_astdb.so Sorcery Astdb Object Wizard 0 Running core
res_sorcery_config.so Sorcery Configuration File Object Wizard 0 Running core
res_sorcery_memory.so Sorcery In-Memory Object Wizard 0 Running core
res_sorcery_memory_cache.so Sorcery Memory Cache Object Wizard 0 Running core
res_sorcery_realtime.so Sorcery Realtime Object Wizard 0 Running core
res_speech.so Generic Speech Recognition API 0 Running core
res_stasis.so Stasis application support 2 Running core
res_stasis_answer.so Stasis application answer support 0 Running core
res_stasis_device_state.so Stasis application device state support 0 Running core
res_stasis_playback.so Stasis application playback support 0 Running core
res_stasis_recording.so Stasis application recording support 0 Running core
res_stasis_snoop.so Stasis application snoop support 0 Running core
res_timing_pthread.so pthread Timing Interface 0 Running extended
res_timing_timerfd.so Timerfd Timing Interface 0 Running core
185 modules loaded
asterisk1-dev*CLI>
I also checked to make sure sound file exist.
asterisk1-dev logrotate.d # ls -l /var/lib/asterisk/sounds/ja/ | egrep "all-c"
-rw-r--r-- 1 asterisk asterisk 13862 Jan 20 2016 all-circuits-busy-now.alaw
-rw-r--r-- 1 asterisk asterisk 13863 Jan 20 2016 all-circuits-busy-now.g722
-r--r--r-- 1 asterisk asterisk 1730 Jan 20 2016 all-circuits-busy-now.g729
-rw-r--r-- 1 asterisk asterisk 2871 Jan 20 2016 all-circuits-busy-now.gsm
-rw-r--r-- 1 asterisk asterisk 10320 Jan 20 2016 all-circuits-busy-now.siren14
-r--r--r-- 1 asterisk asterisk 6880 Jan 20 2016 all-circuits-busy-now.siren7
-rw-r--r-- 1 asterisk asterisk 55450 Jan 20 2016 all-circuits-busy-now.sln16
-rw-r--r-- 1 asterisk asterisk 13862 Jan 20 2016 all-circuits-busy-now.ulaw
-rw-r--r-- 1 asterisk asterisk 27768 Jan 20 2016 all-circuits-busy-now.wav