No SIP commands in the CLI


Fresh install of AsteriskNOW Beta.

No SIP related commands available in the CLI.

I have added two SIP extensions (that don’t work) and rebooted/reinstalled several times to no avail.

Any ideas?


Please explain what you mean by sip related commands ? What are you looking to do ?

sip show peers
"No such command ‘sip show peers’ (type ‘help sip show’ for other possible commands)"
sip show modules
"No such command ‘sip show modules’ (type ‘help sip show’ for other possible commands)“
help sip
"No such command ‘sip’.”

What version of asterisk ?

The version that comes with the AsteriskNOW Beta.


There is no version of asterisk “”. Most likely 1.4.22. Have a look at your log files and see if asterisk is having an issue accessing core files on start.

Sorry, that was FreePBX version.
Asterisk is 1.4.22

Which log file should I look at?


[root@static-host asterisk]# cd /var/log/asterisk/

[root@static-host asterisk]# grep sip full
[Dec 5 17:19:12] NOTICE[2842] chan_sip.c: Unable to load config sip.conf

[root@static-host asterisk]# cd /etc/asterisk

[root@static-host asterisk]# ls -l sip.conf
lrwxrwxrwx 1 apache apache 45 Dec 5 17:21 sip.conf -> /var/www/html/admin/modules/core/etc/sip.conf

[root@static-host asterisk]# ls -l /var/www/html/admin/modules/core/etc/sip.conf
-rw-r–r-- 1 apache apache 2162 Oct 20 15:28 /var/www/html/admin/modules/core/etc/sip.conf

[root@static-host asterisk]# cat sip.conf
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: ;


; These files will all be included in the [general] context
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
; jbenable=yes
; jbforce=yes
;It is also the proper place to add the lines needed for sip nat’ing when going
;through a firewall. For nat’ing you’d need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line 1000 in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
#include sip_custom_post.conf
[root@static-host asterisk]#