[asterisk-users] SIP OPTIONS dialog not understood

[Steve Edwards]

I'm (still) trying to get my Asterisk box talking to a Metaswitch. All I'm 
getting is a "heartbeat" of OPTIONS messages coming from the Metaswitch 
which my Asterisk box replies to. The exchange looks like:

<-- SIP read from 172.b.c.d:5060:
OPTIONS sip:metaswitch at 206.b.c.d:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 
172.b.c.d:5060;rport;branch=z9hG4bK-17eb587208b656d9c2fbd516b5e5401e-172.b.c.d-1
Allow-Events: message-summary
Allow-Events: refer
Allow-Events: dialog
Allow-Events: line-seize
Max-Forwards: 70
Call-ID: BECE8AC6 at 172.b.c.d
From: 
<sip:metaswitch at 172.b.c.d:5060;transport=udp>;tag=172.b.c.d+1+0+22022a3b
CSeq: 445762257 OPTIONS
Organization: Supported: 100rel
Content-Length: 0
Contact: <sip:metaswitch at 172.b.c.d:5060;transport=udp>
To: <sip:metaswitch at 206.b.c.d>


--- (15 headers 0 lines) ---
Looking for metaswitch in test (domain 206.b.c.d)
Transmitting (no NAT) to 172.b.c.d:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
172.b.c.d:5060;rport;branch=z9hG4bK-17eb587208b656d9c2fbd516b5e5401e-172.b.c.d-1;received=172.b.c.d
From: 
<sip:metaswitch at 172.b.c.d:5060;transport=udp>;tag=172.b.c.d+1+0+22022a3b
To: <sip:metaswitch at 206.b.c.d>;tag=as6a59273b
Call-ID: BECE8AC6 at 172.b.c.d
CSeq: 445762257 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:206.b.c.d>
Accept: application/sdp
Content-Length: 0

Is this how OPTIONS is supposed to look? One thing that struck me as 
curious is that I had to add an extension "metaswitch" to my "test" 
context in my dialplan. Otherwise I got "404's."

Can anybody explain (or point to an explanation)?

Thanks in advance,
------------------------------------------------------------------------
Steve Edwards      sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                             Fax: +1-760-731-3000