hello every body.
i want to batch register into asterisk (v 1.4.2) in use sipp (v 3.2). but when
coming OPTIONS message from asterisk, the sipp can not handle it.
Options__________________________________________________________________________________________________________________________
OPTIONS sip:3007@10.8.103.216:5060 SIP/2.0
Via: SIP/2.0/UDP 10.8.103.254:5060;branch=z9hG4bK36df69a8;rport
From: "Unknown" <sip:Unknown@10.8.103.254:5060>;tag=as3a872d19
To: <sip:3007@10.8.103.216:5060>
Contact: <sip:Unknown@10.8.103.254>
Call-ID: 149396b64ce3ff4824d9a91d4adb6865@10.8.103.254:5060
CSeq: 102 OPTIONS
User-Agent: HOLLYUC PBX
Max-Forwards: 70
Date: Thu, 23 Sep 2010 06:17:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
my reg.xml
------------------------------------------------
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="Basic Sipstone UAC">
<send retrans="1000">
<![CDATA[
REGISTER sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
Contact: <sip:[field0]@[local_ip]:[local_port]>
Max-Forwards: 70
From: "[field0]"<sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
To: "[field0]"<sip:[field0]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 REGISTER
Expires: 3600
Content-Length: 0
]]>
</send>
<recv response="100" ></recv>
<recv response="401" auth="true"></recv>
<send retrans="500">
<![CDATA[
REGISTER sip:[remote_ip] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
Max-Forwards: 70
Contact: <sip:[field0]@[local_ip]:[local_port]>
To: "[field0]"<sip:[field0]@[remote_ip]>
From: "[field0]"<sip:[field0]@[remote_ip]:[local_port]>;tag=[call_number]
Call-ID: [call_id]
CSeq: 2 REGISTER
Expires: 3600
[field1]
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,NOTIFY,MESSAGE,SUBSCRIBE,INFO
Content-Length: 0
]]>
</send>
<recv response="100" optional="true"></recv>
<recv response="200" optional="true"></recv>
<label id= "1" />
<recv request="102" timeout="10000" ontimeout="1"></recv>
<send>
<![CDATA[
SIP/2.0 200 OK
Via: SIP/2.0/[transport] [remote_ip]:[remote_port]
Contact: sip:sipp@[local_ip]:[local_port]
To: sut <sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
From: sipp <sip:sipp@[remote_ip]:[remote_port]>;tag=[call_number]
Call-ID: [call_id]
CSeq: 102 OPTIONS
Max-Forwards: 70
Accept: application/sdp
Content-Length: 0
]]>
</send>
<recv response="100" optional="true"></recv>
<recv response="180" optional="true"></recv>
<recv response="200" optional="true"></recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true"></recv>
<!--
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
[authentication username=foouser]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
t=0 0
c=IN IP[media_ip_type] [media_ip]
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
-->
</scenario>
2012-01-10
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