Hi Andre,

The problem is that you are trying to put 2 scenarios into one:
registration followed by an incoming INVITE. However these will have
different Call-IDs. Therefore SIPp will not handle the incoming INVITE
as par of the same scenario, and also your scenario is a client side one
(starts with <send>) so SIPp also cannot start a new call for the
received INVITE, hence the error.

One possible solution is to separate the 2 scenarios and run them
separately one after the other.

Hope this helps.
-David

-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 07 May 2007 20:39
To: sipp-users@lists.sourceforge.net
Subject: [Sipp-users] recv INVITE

Hi all

I wrote a scenario which looks as follows:

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="Asterisk Registration">

  <send retrans="1000">
    <![CDATA[
  REGISTER sip:192.168.1.35 SIP/2.0
  Via: SIP/2.0/UDP 192.168.1.35:[local_port]
  Max-Forwards:5
  To:123456<sip:[EMAIL PROTECTED]:[local_port]>
  From: 123456<sip:[EMAIL PROTECTED]:[local_port]>; tag=123456
  Call-ID: [call_id]
  Cseq: 1 REGISTER
  Contact: <sip:[EMAIL PROTECTED]:[local_port]>
  Expires: 7200
  Content-Length:0
    ]]>
  </send>

  <recv response="100" optional="true">
  </recv>

  <recv response="200">
  </recv>

  <recv request="INVITE">
  </recv>

</scenario>

I start this with ./sipp -sf register.xml -p 5080 -m 1 -affnr 12346 -mp 
6071 192.168.1.35 and everything works fine including response=200. 
Then I INVITE my client from Asterisk but the client can not match the 
recieved INVITE package to the xml scenario and gives the following 
error message:

2007-05-07 20:24:58: Discarding message which can't be mapped to a 
known SIPp call:
INVITE sip:[EMAIL PROTECTED]:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.35:5060;branch=z9hG4bK52375152;rport
From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as3068dc59
To: <sip:[EMAIL PROTECTED]:5080>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 07 May 2007 18:24:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 490

v=0
o=root 2985 2985 IN IP4 192.168.1.35
s=session
c=IN IP4 192.168.1.35
t=0 0
m=audio 17850 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


What's the mistake here? Do I have to set more parameters in the recv 
statement or what is the problem. Thanks for your help.

Andre


------------------------------------------------------------------------
-
This SF.net email is sponsored by DB2 Express
Download DB2 Express C - the FREE version of DB2 express and take
control of your XML. No limits. Just data. Click to get it now.
http://sourceforge.net/powerbar/db2/
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users

-------------------------------------------------------------------------
This SF.net email is sponsored by DB2 Express
Download DB2 Express C - the FREE version of DB2 express and take
control of your XML. No limits. Just data. Click to get it now.
http://sourceforge.net/powerbar/db2/
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users

Reply via email to