Call forwarding (Follow Me) to a mobile phone does not work

Hello!

Please help solve the problem. Thanks.

Logs:

[2020-04-15 12:26:34] WARNING[14666]: file.c:663 ast_openstream_full: File cannot-complete-as-dialed does not exist in any format
[2020-04-15 12:26:34] WARNING[14666]: file.c:958 ast_streamfile: Unable to open cannot-complete-as-dialed (format 0x4 (ulaw)): No such file or directory
[2020-04-15 12:26:34] WARNING[14666]: app_playback.c:475 playback_exec: ast_streamfile failed on Local/[email protected];2 for silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer
[2020-04-15 12:26:34] WARNING[14666]: file.c:663 ast_openstream_full: File check-number-dial-again does not exist in any format
[2020-04-15 12:26:34] WARNING[14666]: file.c:958 ast_streamfile: Unable to open check-number-dial-again (format 0x4 (ulaw)): No such file or directory
[2020-04-15 12:26:34] WARNING[14666]: app_playback.c:475 playback_exec: ast_streamfile failed on Local/[email protected];2 for silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer

ls -l /usr/share/asterisk/sounds/en/silence/1.gsm
-rw-r–r-- 1 root root 1650 окт. 5 2011 /usr/share/asterisk/sounds/en/silence/1.gsm

PBX Core settings

Version: 1.8.13.1~dfsg-3ubuntu3
Build Options: LOADABLE_MODULES
Maximum calls: Not set
Maximum open file handles: Not set
Verbosity: 0
Debug level: 0
Maximum load average: 0.000000
Minimum free memory: 0 MB
Startup time: 10:02:55
Last reload time: 12:05:57
System: Linux/3.2.0-37-generic built by buildd on x86_64 2013-10-16 23:58:01 UTC
System name:
Entity ID: 0c:c4:7a:01:b4:91
Default language: en
Language prefix: Enabled
User name and group: /
Executable includes: Disabled
Transcode via SLIN: Enabled
Internal timing: Enabled
Transmit silence during rec: Disabled
Generic PLC: Enabled

  • Subsystems

    Manager (AMI): Enabled
    Web Manager (AMI/HTTP): Disabled
    Call data records: Enabled
    Realtime Architecture (ARA): Disabled

  • Directories

    Configuration file:
    Configuration directory: /etc/asterisk
    Module directory: /usr/lib/asterisk/modules
    Spool directory: /var/spool/asterisk
    Log directory: /var/log/asterisk
    Run/Sockets directory: /var/run/asterisk
    PID file: /var/run/asterisk/asterisk.pid
    VarLib directory: /var/lib/asterisk
    Data directory: /usr/share/asterisk
    ASTDB: /var/lib/asterisk/astdb
    IAX2 Keys directory: /usr/share/asterisk/keys
    AGI Scripts directory: /usr/share/asterisk/agi-bin

sudo -u asterisk cat /usr/share/asterisk/sounds/en/silence/1.gsm
▒ ▒▒ZPI$▒I$PI$▒I$PI$▒I$PI$▒I$▒ ▒▒ZPI$▒I$PI$▒I$PI$▒I$PI$▒I$▒ ▒▒ZPI$▒I$PI$▒I$PI$▒I$

ls /usr/lib/asterisk/modules | grep codec

codec_adpcm.so
codec_alaw.so
codec_a_mu.so
codec_dahdi.so
codec_g722.so
codec_g726.so
codec_gsm.so
codec_lpc10.so
codec_speex.so
codec_ulaw.so

HI

Did You enable Follow me and use # after mobile number ?
Can you call directly to that mobile number from your ext ?

Did You enable Follow me and use # after mobile number ?
Yes of course

Can you call directly to that mobile number from your ext ?
Уes, calls to mobile from the extension pass

is this a freepbx system? How did you install it?

FreePBX 2.10.1.19 + Asterisk 1.8.13.1 + Ubuntu 13.10
It was installed a long time ago, not by me

and what is the content of /etc/asterisk/asterisk.conf ?

[directories](!)

astetcdir => /etc/asterisk
astmoddir => /usr/lib/asterisk/modules
astvarlibdir => /var/lib/asterisk
astdbdir => /var/lib/asterisk
astkeydir => /var/lib/asterisk
astdatadir => /usr/share/asterisk
astagidir => /usr/share/asterisk/agi-bin
astspooldir => /var/spool/asterisk
astrundir => /var/run/asterisk
astlogdir => /var/log/asterisk

[options]
;verbose = 3
;debug = 3
;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.
;internal_timing = yes
;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.
;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_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.

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

[compat]
pbx_realtime=1.6
res_agi=1.6
app_set=1.6

I think the problem is that asterisk cannot find the file “cannot-complete-as-dialed (format 0x4 (ulaw))” and “cannot-complete-as-dialed”, but I don’t know how to check it, by what paths these files should be located.

in your “astdatadir” subdirectory en

Not an answer to your question, but this system went EOL 7-8 years ago. You need a plan to get on something supportable.

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