Thanks for your reply, 
 
In fact, it works great with xlite, but I really need to make it work in my 
way. I'm sure it possible but I missed something !
The question is, what is it ?
 
maybe because off -aa option which crash sipp ? Is somebody already tested this 
possibility ?
Ruhi ASLAN
Stagiaire ST40 - NOC/Operation
 

VTX SERVICES SA
Une société du groupe VTX Telecom
================================================================
Tél. direct : 021 721 12 18
Av. de Lavaux 101 - 1009 Pully
http://www.vtx.ch <http://www.vtx.ch/>  - ruhi.as...@vtx-telecom.ch
----------------------------------------------------------------
VTX, votre partenaire telecom proche de vous !
================================================================
 

________________________________

De : ritesh.gu...@bt.com [mailto:ritesh.gu...@bt.com] 
Envoyé : vendredi, 9. avril 2010 17:51
À : Ruhi Aslan; sipp-users@lists.sourceforge.net
Objet : RE: crazy problem on simple call scenario



Find example below. 

 

 If that does not work then try to connect using any other sip phone such as 
xLite and take network traces for registration process then try to map the 
below message. It will definitely work. 

 

Some time you need to supply same [tag]  as part of subscribe message which you 
got during 200ok  response from server.

ßRegistration  (Send)

à 200 ok (Receive Response)

ßSubscribe  (Send) 

 

  <send >

    <![CDATA[

REGISTER sip:10.230.53.225 SIP/2.0

Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport

Max-Forwards: 70

Contact: <sip:te...@10.230.52.50:[local_port];rinstance=df769c75a8bad123>

To: "test2"<sip:te...@10.230.53.225>

From: "test2"<sip:te...@10.230.53.225>;tag=[call_number]

Call-ID: [call_id]

CSeq: [cseq] REGISTER

Expires: 3600

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO

Content-Length: [len]

]]>

  </send>

 

  <recv response="200" crlf="true">

  </recv>

 

  <send >

    <![CDATA[

SUBSCRIBE sip:te...@10.230.53.225 SIP/2.0

Via: SIP/2.0/UDP 10.230.52.50:[local_port];branch=[branch];rport

Max-Forwards: 70

Contact: <sip:te...@10.230.52.50:[local_port]>

To: "test2"<sip:te...@10.230.53.225>

From: "test2"<sip:te...@10.230.53.225>;tag=[call_number]

Call-ID: [call_id]

CSeq: [cseq] SUBSCRIBE

Expires: 300

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO

Event: message-summary

Content-Length: [len]

 

 

]]>

  </send>

 

  <recv response="501" crlf="true">

  </recv>

 

 

 

From: Ruhi Aslan [mailto:ruhi.as...@vtx-telecom.ch] 
Sent: 09 April 2010 16:13
To: sipp-users@lists.sourceforge.net
Subject: [Sipp-users] crazy problem on simple call scenario

 

________________________________

De : Ruhi Aslan 
Envoyé : vendredi, 9. avril 2010 16:56
À : 'sipp-users-requ...@lists.sourceforge.net'
Objet : help

Hi all,

 

Sipp is a great tool and I currently pull my hair out...

 

I have some trouble with a very simple scenario. I even can't make a call to 
sipp registered phone.

I first registered my phone :

 

                  sipp -sf callee_hangup_process_test.xml -inf 
csv/register_client.csv asterisk.ch -trace_err -r1 -m 1

 

## register my sipp phone to get calls


  <send>
    <![CDATA[

 

REGISTER sip:sipproxy SIP/2.0
Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID
From: <sip:4...@mycomputerip>;tag=1
To: <sip:4...@mycomputerip>
Call-ID: 1...@mycomputerip <mailto:1...@mycomputerip> 
CSeq: 1 REGISTER
Contact: *
Max-Forwards: 5
Expires: 0
User-Agent: SIPp/Linux
Content-Length: 0

 

    ]]>
  </send>
  <recv response="404" optional="true" next="1">
  </recv>

 

  <recv response="401" auth="true">
  </recv>

 

******* Register Process *******


  <send retrans="500">
    <![CDATA[

 

REGISTER sip:sipproxy SIP/2.0
Via: SIP/2.0/UDP mycomputerIP:5060;branch=z9hG4bK-ID
From: <sip:4...@mycomputerip>;tag=1
To: <sip:4...@mycomputerip>
Call-ID: 1...@mycomputerip <mailto:1...@mycomputerip> 
CSeq: 1 REGISTER
Contact: *
[AUTHENTICATION LINE]

Max-Forwards: 5
Expires: 0
User-Agent: SIPp/Linux
Content-Length: 0

 

     ]]>


  </send>

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

 

### phone registered, sip show peer 44 tell me it's OK and reachable on 
mycomputerIP

 

 

Then I ask to it to wait until an INVITE comes :


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

 

 

In another window, I make a call with another phone number 43 ( correct 
scenarios and successfully tested )

 

sipp -sf callee_hangup.xml -inf csv/caller.cvs asterisk.ch -trace_err  -r 1 -m 1

 

BUT, callee_hangup_process_test.xml doesn't get the INVITE from 
callee_hangup.xml scenario. 

The crazy thing is that wireshark says that it sends the expected INVITE to 
callee_hangup_process_test.xml ( on the right computer, on the right port ). 
But on my previous INVITE recv request, the count persist on 0 !

 

 

Here the INVITE sended to mycomputerIP (  supposed to make the  INVITE recv 
reauest count up to 1 )

 

INVITE sip:4...@mycomputerip:5060 SIP/2.0
Record-Route: <sip:sipproxy;lr=on;ftag=ftag;vsf=some...;did=...>
Via: SIP/2.0/UDP sipproxy;branch=z9hG4bK-ID2
Via: SIP/2.0/UDP 
asterisk.ch:5060;received=asterisk.ch;branch=z9hG4b-ID;rport=5060
From: "43" <sip:4...@voip.vtx.ch>;tag=as1cf8af76
To: <sip:4...@mycomputerip:5060>
Contact: <sip:4...@asterisk.ch>
Call-ID: call...@asterisk.ch

CSeq: 102 INVITE
User-Agent: voipua
Max-Forwards: 69
Date: Fri, 09 Apr 2010 13:54:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 242
P-hint: outbound

 

v=0
o=root 26199 26199 IN IP4 asterisk.ch
s=session
c=IN IP4 asterisk.ch
t=0 0
m=audio 18150 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 

 

more info : 

 

I already use -aa option for OPTIONS NOTIFY  request, and on the second 
OPTIONS, sipp crash on seg fault  :-\

 

 

 

So where is my mistake ?

 

Ruhi ASLAN
Stagiaire ST40 - NOC/Operation

 


VTX SERVICES SA
Une société du groupe VTX Telecom
================================================================
Tél. direct : 021 721 12 18
Av. de Lavaux 101 - 1009 Pully
http://www.vtx.ch <http://www.vtx.ch/>  - ruhi.as...@vtx-telecom.ch
----------------------------------------------------------------
VTX, votre partenaire telecom proche de vous !
================================================================

 

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