Rejected because extension not found in context 'users'

General Help
1

HI,

I am using the FreePbx distribution and trying to setup the first internal call between two extensions but getting the following error message.

[2014-11-17 15:48:45] NOTICE[18843][C-00000013]: chan_sip.c:25624 handle_request_invite: Call from ‘simon’ (10.40.207.182:5060) to extension ‘adil’ rejected because extension not found in context ‘users’

This is the version of my Asterisk ( FreePBX package)

Asterisk 11.13.1 built by root @ jenkins-builder1.schmoozecom.net on a x86_64 running Linux on 2014-10-20 17:14:32 UTC.

I have used extensions.conf and sip.conf to define my extensions and this is how they look like. Is there anything else that I need to configure.

sip.conf

[adil]
type=friend
context=users
host=dynamic
secret=unsecurepassword
disallow=all
allow=ulaw

[simon]
type=friend
context=users
host=dynamic
secret=unsecurepassword
disallow=all
allow=ulaw

extensions.conf

[from=internal]
exten = 100,1, Answer()
same = n,Wait(1)
same = n, Playback(hello-we-are-closed)
same = n, Hangup()

[users]
exten=>6001,1,Dial(SIP/adil,20)
exten=>6002,2,Dial(SIP/simon,20)

Could someone please suggest what I need to fix.

2

You should run

head -5 /etc/asterisk/sip.conf

and

head -5 /etc/asterisk/extensions.conf

before editing them, then start from the beginning by reading the wiki here.

http://wiki.freepbx.org

3

Hi,
Thanks for help. I removed the empty lines and also removed the first section in extensions.conf. Now I only have the users block.

[from=internal]
exten = 100,1, Answer()
same = n,Wait(1)
same = n, Playback(hello-we-are-closed)
same = n, Hangup()

I still do not see any improvement, restarted asterisk and loaded the dialplan. Could you provide a working example or a brief step by step guide to get internal call working. I think FreePBX does not seem to be easy for beginners.

4

Some times the hardest thing for beginners to do is to follow directions and read documentation. There are literally hundreds of pages of documentation on the FreePBX wiki on how to use and configure FreePBX. A good start might be to read a few of them. Perhaps start with this one

http://wiki.freepbx.org/display/FD/FreePBX+Distro+First+Steps+After+Installation

head /etc/asterisk/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: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;