[Asterisk-Users] Sip Registration Problem

[Karl Brose]

RFC  3261  states:

11.2 Processing of OPTIONS Request

   The response to an OPTIONS is constructed using the standard rules
   for a SIP response as discussed in Section 8.2.6.  The response code
   chosen MUST be the same that would have been chosen had the request
   been an INVITE.  That is, a 200 (OK) would be returned if the UAS is
   ready to accept a call, a 486 (Busy Here) would be returned if the
   UAS is busy, etc.  This allows an OPTIONS request to be used to
   determine the basic state of a UAS, which can be an indication of
   whether the UAS will accept an INVITE request.

   An OPTIONS request received within a dialog generates a 200 (OK)
   response that is identical to one constructed outside a dialog and
   does not have any impact on the dialog.

   This use of OPTIONS has limitations due to the differences in proxy
   handling of OPTIONS and INVITE requests.  While a forked INVITE can
   result in multiple 200 (OK) responses being returned, a forked
   OPTIONS will only result in a single 200 (OK) response, since it is
   treated by proxies using the non-INVITE handling.  See Section 16.7
   for the normative details.

   If the response to an OPTIONS is generated by a proxy server, the
   proxy returns a 200 (OK), listing the capabilities of the server.
   The response does not contain a message body.

   Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
   fields SHOULD be present in a 200 (OK) response to an OPTIONS
   request.  If the response is generated by a proxy, the Allow header
   field SHOULD be omitted as it is ambiguous since a proxy is method
   agnostic.  Contact header fields MAY be present in a 200 (OK)
   response and have the same semantics as in a 3xx response.  That is,
   they may list a set of alternative names and methods of reaching the
   user.  A Warning header field MAY be present.

   A message body MAY be sent, the type of which is determined by the
   Accept header field in the OPTIONS request (application/sdp is the
   default if the Accept header field is not present).  If the types
   include one that can describe media capabilities, the UAS SHOULD
   include a body in the response for that purpose.  Details on the
   construction of such a body in the case of application/sdp are
   described in [13].




Brett Nemeroff wrote:

>How will this effect a live system? No new calls? Or will it terminate
>exisiting calls?
>
>I'll have a chat with the vendor regarding the OPTIONS reply.. It
>certainly does sesem like it should reply with something..
>
>-----Original Message-----
>From: asterisk-users-admin at lists.digium.com
>[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Olle E.
>Johansson
>Sent: Tuesday, May 25, 2004 1:13 AM
>To: asterisk-users at lists.digium.com
>Subject: Re: [Asterisk-Users] Sip Registration Problem
>
>
>Karl Brose wrote:
>
>  
>
>>Btw, Ignoring OPTIONS is not a valid option (:-) whether sip proxy or
>>not, Asterisk doesn't do it correctly either.
>>The host should respond with 200/OK if the call >could< succeed 
>>theoretically if it were an INVITE or else it should send a
>>404 or maybe a 487(? hmm, have to look)  see the RFC for details.
>>    
>>
>Interesting, didn't know that. Where in the RFC?
>
>
>  
>
>>>I removed the qualify lines and sip reload [ed]. The extension still 
>>>showed up as "UNREACHABLE" instead of "UNMONITORED". I had to do a 
>>>full restart to get it to stop sending the OPTIONS messages.
>>> 
>>>What did I do wrong here? How can I make a change to qualify without 
>>>restarting?
>>>      
>>>
>If a peer is registred at reload/sip reload, it will not change.
>You have to unload the sip module and reload it or restart asterisk
>to change the configuration of a registred, i.e. active, peer.
>
>/O
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