[asterisk-dev] Has the meaning of the "s" extension changed recently?

[Enzo Michelangeli]

Wasn't the "s" extension supposed to mean "the null extension"?
http://www.digium.com/asterisk_handbook/extensions.conf.html says:

s: Defines how to route a call when no other routing information has been
received. On a PRI or local FXS line, we will receive a number string to
route the call by. When receiving a call from an analog (FXO) line, we
won't get any routing information, just a ring. In this case the 's'
extension will be used.

Also http://www.voip-info.org/wiki/index.php?page=Asterisk+s+extension
says similar things, and that was consistent with the behaviour of
Asterisk 1.0.x .

However, I recently noticed that SIP INVITEs sent to a null user (e.g.
"INVITE sip:jaxhk.hn.org SIP/2.0", generated by a
"Dial(SIP/@jaxhk.hn.org)" in the dialplan) do not trigger the
"exten => s,..." lines in the receiving peer (running * 1.2.4). Instead,
the latter appears to look for a _user_ equal to the domain, in an empty
domain:

   Looking for jaxhk.hn.org in default (domain )

Everything works if I use "Dial(SIP/s at jaxhk.hn.org)", but this begs the
question of which pattern I should use for an empty username...

Enzo

P.S. Below is the "sip debug".

burning*CLI>
<-- SIP read from 218.103.207.95:5060:
INVITE sip:jaxhk.hn.org SIP/2.0
Via: SIP/2.0/UDP 218.103.207.95:5060;branch=z9hG4bK4ae032b0
From: "Enzo Michelangeli" <sip:+85227109517 at 218.103.207.95>;tag=as11610026
To: <sip:jaxhk.hn.org>
Contact: <sip:+85227109517 at 218.103.207.95>
Call-ID: 3702a90d19a5c90a6d35612103d9be13 at 218.103.207.95
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 02 Feb 2006 15:44:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 294

v=0
o=root 25983 25983 IN IP4 218.103.207.95
s=session
c=IN IP4 218.103.207.95
t=0 0
m=audio 16128 RTP/AVP 0 8 3 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (12 headers 13 lines)---
Using INVITE request as basis request -
3702a90d19a5c90a6d35612103d9be13 at 218.103.207.95
Sending to 218.103.207.95 : 5060 (non-NAT)
Found no matching peer or user for '218.103.207.95:5060'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 2
Found RTP audio format 101
Peer audio RTP is at port 218.103.207.95:16128
Found description format PCMU
Found description format PCMA
Found description format GSM
Found description format G726-32
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x1e
(gsm|ulaw|alaw|g726)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for jaxhk.hn.org in default (domain )
Reliably Transmitting (no NAT) to 218.103.207.95:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
218.103.207.95:5060;branch=z9hG4bK4ae032b0;received=218.103.207.95
From: "Enzo Michelangeli" <sip:+85227109517 at 218.103.207.95>;tag=as11610026
To: <sip:jaxhk.hn.org>;tag=as47a8b4cf
Call-ID: 3702a90d19a5c90a6d35612103d9be13 at 218.103.207.95
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:jaxhk.hn.org at 203.198.106.87>
Content-Length: 0