________________________________
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
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http://www.vtx.ch <http://www.vtx.ch/> - ruhi.as...@vtx-telecom.ch
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