Hi,all

i want to continuity hold more than one calls from other agents, but use single 
following scenario  xml file..
just can hold only one call.

regards.
------------------------------------------
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="Basic UAS responder">
  <recv request="INVITE" crlf="true">
  </recv>
  <send>
    <![CDATA[
      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0
    ]]>
  </send>
  <send retrans="500">
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Type: application/sdp
      Content-Length: [len]
      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000
    ]]>
  </send>
  <recv request="ACK"
        optional="true"
        rtd="true"
        crlf="true">
  <!--<action>
    <exec play_pcap_audio="g711a.pcap"/>
  </action>-->
  </recv>
  <recv request="BYE">
  </recv>
  <send>
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0
    ]]>
  </send>
   <pause milliseconds="2000"></pause>
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>



2012-03-08 



raojlist 



发件人: raojlist 
发送时间: 2012-03-07  10:24:13 
收件人: sipp-users 
抄送: 
主题: [Sipp-users] Hello all, I want to batch register agents ,and then hold 
incoming call, but have problem. 
 
Hello all,

I can register into asterisk successful,but can not hold incoming call from 
other agent client registered by eyebeam,
my sipp xml config file describes. following:




<?xml version="1.0" encoding="us-ascii"?>
<scenario name="New_Call">
 <send retrans="500">
  <![CDATA[
REGISTER sip:[field0]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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
Contact: sip:[field0]@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Expires:3600
Content-Length: 0

]]>
 </send>
 <recv response="100" optional="true" />
 <recv response="200" crlf="true" />
 <send>
  <![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0

]]>
 </send>
 <recv request="INVITE" crlf="true" />
 <send>
  <![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]

v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000]]>
 </send>
 <recv request="ACK" crlf="true" />
 <pause milliseconds="13000" />
 <send retrans="500">
  <![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

]]>
 </send>
 <label id="1" />
 <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200" />
 <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000" />
</scenario>
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