I'm an asterisk newbie and I've just installed it on my ubuntu 12.04 server with sudo apt-get install asterisk. I'm following the tutorial for asterisk from here: https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality
which describes setting up the most simplest PBX with two sip phones demo-alice and demo-bob. I've followed the instruction to the dot, however, when I make a call from demo-bob to demo-alice I get:
[Feb 5 13:23:03] NOTICE[13667]: chan_sip.c:22622 handle_request_invite:
Call from 'demo-bob' (192.168.1.2:5060) to extension '6001' rejected
because extension not found in context 'users'.
However, the dialplan gives me:
*CLI> dialplan show users
[ Context 'users' created by 'pbx_config' ]
'6001' => 1. Dial(SIP/demo-alice) [pbx_config]
'6002' => 1. Dial(SIP/demo-bob) [pbx_config]
-= 2 extensions (2 priorities) in 1 context. =-
So I clearly HAVE extension 6001 in context users. What am I doing wrong? Please help?
Some more info:
I've reloaded the config and dialplan with sip reload and dialplan reload. I'm using asterisk 1.8. This is the output of sip show peers:
*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
demo-alice/demo-alice 192.168.1.12 D N A 5060 Unmonitored
demo-bob/demo-bob 192.168.1.2 D N A 5060 Unmonitored
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 2 online, 0 offline]