[asterisk-users] When shall SIP phone reply "480 Temporarily Unavailable"

[Olivier]

Hello,

When sending this AMI request ...
192.168.64.5 -> Action: Originate
192.168.64.5 -> Channel: SIP/9122
192.168.64.5 -> Async: True
192.168.64.5 -> Callerid: 9122 Guest2 <9122>
192.168.64.5 -> Exten: 9123
192.168.64.5 -> Context: local
192.168.64.5 -> Priority: 1

... I've got this INVITE from Asterisk
INVITE sip:9121 at 192.168.100.198:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport
From: "9121 Guest1" <sip:9121 at 192.168.100.254 <sip%3A9121 at 192.168.100.254>
>;tag=as237a9159
To: <sip:9121 at 192.168.100.198:5060;user=phone>
Contact: <sip:9121 at 192.168.100.254 <sip%3A9121 at 192.168.100.254>>
Call-ID: 6bddeb200c2aee553856dab4098c6f8e at 192.168.100.254
CSeq: 102 INVITE
User-Agent: Asterisk PBX



As you can see, SIP From and To headers are different but both somehow refer
to the peer.
When receiving such INVITE, my SIP hardphone (a Thomson ST2030) replies with
:

SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport
From: "9121 Guest1"<sip:9121 at 192.168.100.254 <sip%3A9121 at 192.168.100.254>
>;tag=as237a9159
To: <sip:9121 at 192.168.100.198:5060;user=phone>;tag=c0a80101-a611e
Call-ID: 6bddeb200c2aee553856dab4098c6f8e at 192.168.100.254
CSeq: 102 INVITE
Content-Length: 0




Is it normal to reply this way ?
I tried with another SIP phone (a Siemens Gigaset) and it accepted the
INVITE (and started to ring).

regards
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