As far as I understand SIP Regisration procedure, the problem is Contact: *
I would like to suggest you to put a meaningful SIP URI there.
Regards,
Evgeny Miloslavsky
Systest Engineer
Juniper Networks Solutions Israel LTD.
Office: 972-9-9712355
Office: 972-74-7170072
________________________________
From: Ruhi Aslan [mailto:ruhi.as...@vtx-telecom.ch]
Sent: Friday, April 09, 2010 7:01 PM
To: sipp-users@lists.sourceforge.net
Subject: [Sipp-users] TR: crazy problem on simple call scenario
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.
<--Registration (Send)
--> 200 ok (Receive Response)
<--Subscribe (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
<sip:te...@10.230.52.50:%5blocal_port%5d;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]
<sip:te...@10.230.52.50:%5blocal_port%5d> >
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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