Asterisk 1.6.0.6 : Call from 'x' to extension 'x' rejected

Asterisk Asterisk Support
1

Hello,
I have a basic setup, with only two SIP phones declared in my sip.conf, and an extension.conf dialplan including those two SIP, an echo service, and a “playback” service.
However, when I try to call 1001 from 1000, or the oposite, or calling the 1002 service, I always have a message of the form :
[Apr 6 18:50:44] NOTICE[25637] chan_sip.c: Call from ‘1000’ to extension ‘1001’ rejected because extension not found.

This is because of the lacking “@domain” entered in my SIP phone : indeed, I want to be able to call directly the extension, with “SIP:1002” for example, and not “SIP:1002@something”.

Is there a way not to have to enter a domain for every local call ?

Thanks by advance, below are my conf files.

============= sip.conf ===============
[general]
context=internal
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[1000]
type=friend
context=internal
host=dynamic

[1001]
type=friend
context=internal
host=dynamic

============= extension.conf ===============

[globals]

[general]
autofallthrough=yes

[default]

[incoming_calls]

[internal]
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()

exten => 1000,1,Verbose(1|Extension 1000)
exten => 1000,n,Dial(SIP/1000,30)
exten => 1000,n,Hangup()

;exten => 1001,1,Verbose(1|Extension 1001)
exten => 1001,1,Dial(SIP/1001,30)
exten => 1001,n,Hangup()

exten => 1002,1,Answer()
exten => 1002,n,Playback(/var/lib/asterisk/sounds/en/agent-loggedoff.gsm)
exten => 1002,n,Hangup()

[phones]
include => internal

2

One would not normally have to supply a domain when dialling, although the phone should always add one. Phones would normally be configured so that you didn’t need the SIP:, either.

I think your diagnosis is wrong and you need to turn up the tracing.

The Book doesn’t mention context in the general section, so I wonder if your phones are actually be treated as anonymous.