How to use call files with Asterisk 16 running on Ubuntu 22.04

Please let me know if I am in the wrong place.

I need to make SIP calls using call files.
Currently - On FreePBX versions the directory to use was at /var/spool/asterisk/outgoing/ and I could simply drop in my call file and the call would be processed.

New - I installed Ubuntu 22.04 and Asterisk 16.29.0 and the outgoing directory is no longer there. I checked to be sure pbx_config.so shows it is running.
module show
pbx_config.so Text Extension Configuration 0 Running core

I used the sample /etc/asterisk/asterisk.conf file that contains the location of where I expected to find the outgoing directory. I also added an /outgoing/ directory to /var/spool/asterisk/

After this a rebooted the server but I am not able to get calls to process by dropping them into the /var/spool/asterisk/outgoing/ directory.

Below is the data in /etc/asterisk/asterisk.conf

[directories]
astcachedir => /tmp
astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /var/lib/asterisk
astagidir => /var/lib/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk
astsbindir => /usr/sbin

[options]
;verbose = 3
;debug = 3
;trace = 0 ; Set the trace level.
;refdebug = yes ; Enable reference count debug logging.
;alwaysfork = yes ; Same as -F at startup.
;nofork = yes ; Same as -f at startup.
;quiet = yes ; Same as -q at startup.
;timestamp = yes ; Same as -T at startup.
;execincludes = yes ; Support #exec in config files.
;console = yes ; Run as console (same as -c at startup).
;highpriority = yes ; Run realtime priority (same as -p at
; startup).
;initcrypto = yes ; Initialize crypto keys (same as -i at
; startup).
;nocolor = yes ; Disable console colors.
;dontwarn = yes ; Disable some warnings.
;dumpcore = yes ; Dump core on crash (same as -g at startup).
;languageprefix = yes ; Use the new sound prefix path syntax.
;systemname = my_system_name ; Prefix uniqueid with a system name for
; Global uniqueness issues.
;autosystemname = yes ; Automatically set systemname to hostname,
; uses ‘localhost’ on failure, or systemname if
; set.
;mindtmfduration = 80 ; Set minimum DTMF duration in ms (default 80 ms)
; If we get shorter DTMF messages, these will be
; changed to the minimum duration
;maxcalls = 10 ; Maximum amount of calls allowed.
;maxload = 0.9 ; Asterisk stops accepting new calls if the
; load average exceed this limit.
;maxfiles = 1000 ; Maximum amount of openfiles.
;minmemfree = 1 ; In MBs, Asterisk stops accepting new calls if
; the amount of free memory falls below this
; watermark.
;cache_media_frames = yes ; Cache media frames for performance
; Disable this option to help track down media frame
; mismanagement when using valgrind or MALLOC_DEBUG.
; The cache gets in the way of determining if the
; frame is used after being freed and who freed it.
; NOTE: This option has no effect when Asterisk is
; compiled with the LOW_MEMORY compile time option
; enabled because the cache code does not exist.
; Default yes
;cache_record_files = yes ; Cache recorded sound files to another
; directory during recording.
;record_cache_dir = /tmp ; Specify cache directory (used in conjunction
; with cache_record_files).
;transmit_silence = yes ; Transmit silence while a channel is in a
; waiting state, a recording only state, or
; when DTMF is being generated. Note that the
; silence internally is generated in raw signed
; linear format. This means that it must be
; transcoded into the native format of the
; channel before it can be sent to the device.
; It is for this reason that this is optional,
; as it may result in requiring a temporary
; codec translation path for a channel that may
; not otherwise require one.
;transcode_via_sln = yes ; Build transcode paths via SLINEAR, instead of
; directly.
;runuser = asterisk ; The user to run as.
;rungroup = asterisk ; The group to run as.
;lightbackground = yes ; If your terminal is set for a light-colored
; background.
;forceblackbackground = yes ; Force the background of the terminal to be
; black, in order for terminal colors to show
; up properly.
;defaultlanguage = en ; Default language
documentation_language = en_US ; Set the language you want documentation
; displayed in. Value is in the same format as
; locale names.
;hideconnect = yes ; Hide messages displayed when a remote console
; connects and disconnects.
;lockconfdir = no ; Protect the directory containing the
; configuration files (/etc/asterisk) with a
; lock.
;stdexten = gosub ; How to invoke the extensions.conf stdexten.
; macro - Invoke the stdexten using a macro as
; done by legacy Asterisk versions.
; gosub - Invoke the stdexten using a gosub as
; documented in extensions.conf.sample.
; Default gosub.
;live_dangerously = no ; Enable the execution of ‘dangerous’ dialplan
; functions from external sources (AMI,
; etc.) These functions (such as SHELL) are
; considered dangerous because they can allow
; privilege escalation.
; Default no
;entityid=00:11:22:33:44:55 ; Entity ID.
; This is in the form of a MAC address.
; It should be universally unique.
; It must be unique between servers communicating
; with a protocol that uses this value.
; This is currently is used by DUNDi and
; Exchanging Device and Mailbox State
; using protocols: XMPP, Corosync and PJSIP.
;rtp_use_dynamic = yes ; When set to “yes” RTP dynamic payload types
; are assigned dynamically per RTP instance vs.
; allowing Asterisk to globally initialize them
; to pre-designated numbers (defaults to “yes”).
;rtp_pt_dynamic = 35 ; Normally the Dynamic RTP Payload Type numbers
; are 96-127, which allow just 32 formats. The
; starting point 35 enables the range 35-63 and
; allows 29 additional formats. When you use
; more than 32 formats in the dynamic range and
; calls are not accepted by a remote
; implementation, please report this and go
; back to value 96.
;hide_messaging_ami_events = no; This option, if enabled, will
; suppress all of the Message/ast_msg_queue channel’s
; housekeeping AMI and ARI channel events. This can
; reduce the load on the manager and ARI applications
; when the Digium Phone Module for Asterisk is in use.

; Changing the following lines may compromise your security.
;[files]
;astctlpermissions = 0660
;astctlowner = root
;astctlgroup = apache
;astctl = asterisk.ctl

Does ‘core show settings’ yield any clues with the ‘Spool directory’ setting?

Does ‘module show like spool’ show that ‘pbx_spool.so’ is loaded?

If you bump up console logging does anything relevant show on the console when you copy a file to the outgoing spool directory?

sedwards, thanks for the reply. I was able to see that it was working once I turned verbosity and debug up. Now to get an RTP engine setup. Have a great day. :slight_smile:

What RTP engine? For what?

Before you go much further, please note that mainline support for Asterisk 16 ended over a month ago. You should be using Asterisk 18 or 20.