Jeff,

I can see in the message file that the correct messages are being received. Could you describe the issue further in detail

Also If I understand correctly,

1. You register a phone with id 4501 from a pc using a sipp script.
2. You register a phone with id 4502 from the same pc again using sipp
3. you call towards 4502 using a sipp script using the same script.
4. you want to receive a call for 4502 using another sipp script.

Here's what you need to be careful about.

You need to run the server script to receive a call on the same port that was specified in the register message for 4502.
Any invite coming from the registrar would follow the contact in the register message.

also please use the following options in your commands

sipp 10.0.60.1 -sf 4501reg.xml -m 1 -mp 4501 -i <local_ip> -p <port> when sending the registers.


Thanks,
Makarand


On 10/27/06, Jeff Ollier <[EMAIL PROTECTED]> wrote:
The commands are:
 
sipp 10.0.60.1 -sf 4501reg.xml -m 1 -mp 4501 -trace_msg
sipp 10.0.60.1 -sf 4502reg.xml -m 1 -mp 4501 -trace_msg
sipp 10.0.60.1 -sf 4501call4502.xml -m 1 -mp 4501 -trace_msg
 
The invite is sent to 10.0.60.100 (where SIPp resides), but no response to invite.
The 3 commands are sent from the same PC.
 

Jeff OllierNetwork Quality Control Engineer



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-----Original Message-----
From: makarand bhagwat [mailto:[EMAIL PROTECTED]]
Sent: Friday, October 27, 2006 8:39 AM
To: Jeff Ollier
Cc: [email protected]
Subject: Re: [Sipp-users] Invite and Answer

Could you send the message trace for the register sequence as well as the call sequence ?

On 10/26/06, Jeff Ollier < [EMAIL PROTECTED]> wrote:
I am having trouble with this response.  I am trying to register two UAs to the SIP Server (successful) and make calls between them through the SIP Server (unsuccessful). 
 
Invite Scenario:
 
<?xml version="1.0" encoding="ISO-8859-1" ?>
 <scenario name="register_answer">
   <label id="5" />
<send>
<![CDATA[
 
      INVITE sip:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sip:[EMAIL PROTECTED]:[local_port];tag=[call_number]
      To: sip:[EMAIL PROTECTED]:[remote_port]
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 70
      Content-Type: application/sdp
      Content-Length: [len]
 
      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=test
      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 response="200">
  </recv>
<send>
    <![CDATA[
 
      ACK sip:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip];branch=[branch]
      From: "4501" <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
      To: <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      Cseq: 1 ACK
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 70
      Content-Length: 0
 
    ]]>
  </send>
<pause milliseconds="8000"/>
<send>
<![CDATA[
 
BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
 From: "4501"<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
 To: <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
 Call-ID: [call_id]
 CSeq: 2 BYE
 Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Content-Length: 0
 
 ]]>
</send>
 
<recv response="200">
</recv>
 
</scenario>
 
Response scenario:
 
<?xml version="1.0" encoding="ISO-8859-1" ?>
- <scenario name="Basic UAS responder">
  <recv request="INVITE" crlf="true" />
- <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" />
  <recv request="BYE" />
- <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="4000" />
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200" />
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000" />
  </scenario>
 

Jeff OllierNetwork Quality Control Engineer


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-------------------------------------------------------------------------
Using Tomcat but need to do more? Need to support web services, security?
Get stuff done quickly with pre-integrated technology to make your job easier
Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642
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