yes，thanks！
i test use -aa in command line ，actually，it work。 and the peer‘s status is
ok （200）from asterisk ！so register successful。
if i want to start an ooc scenario in coming days，maybe need “the Accept:
application/sdp". but I have not tested it.
2012-01-10
raojlist
发件人： mayamatakeshi
发送时间： 2012-01-10 16:27:58
收件人： Tu Le
抄送： Sipp
主题： Re: [Sipp-users] [Discarding message which can't be mapped to aknown SIPp
call:] Re: Re: Re: some question about sipp handle OPTIONS(102)message.
On Tue, Jan 10, 2012 at 5:15 PM, Tu Le <l...@tma.com.vn> wrote:
It seems your OPTION message have difference Call-ID with your REGISTER. Could
you try to enable auto-answer by using �Caa in command line?
Ah yes, This works. However, this just causes sipp to send a stock "200 OK".
So as long as raojlist doesn't need to send a specific reply (ooc scenario
using option -oocsf), it should work.
But if asterisk is expecting "Accept: application/sdp" in the reply, to be able
to start an ooc scenario will still be required.
From: mayamatakeshi [mailto:mayamatake...@gmail.com]
Sent: Tuesday, January 10, 2012 3:01 PM
To: raojlist
Cc: sipp-users
Subject: Re: [Sipp-users] [Discarding message which can't be mapped to a known
SIPp call:] Re: Re: Re: some question about sipp handle OPTIONS(102) message.
2012/1/10 raojlist <raojl...@163.com>
2012-01-10 15:15:20:867 1326179720.867330: Discarding message which
can't be mapped to a known SIPp call:
OPTIONS sip:3007@10.8.103.216:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.103.254:5060;branch=z9hG4bK0f27650b;rport
From: "Unknown" <sip:Unknown@10.8.103.254:5060>;tag=as1e37f103
To: <sip:3007@10.8.103.216:5060>
Contact: <sip:Unknown@10.8.103.254>
Call-ID: 003e344d599cf73a555321ef1bed3a21@10.8.103.254:5060
CSeq: 102 OPTIONS
User-Agent: HOLLYUC PBX
Max-Forwards: 70
Date: Thu, 23 Sep 2010 07:14:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
I actually never used ooc scenario.
I tried now and confirmed the message is not handled by the ooc scenario.
So i am not sure if this never worked or if it got broken in the latest SIPp
that I have built from source.
You could try this alternative:
http://permalink.gmane.org/gmane.comp.telephony.sipp.user/5769
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