I got the solution .. ... first run the UAC (registration) once registration are done then exit this SIPp instance.
Then run new instance for UAS (Receiving Invite). Now it will work. From: Himanshu Rawat [mailto:himanshu.ra...@gmail.com] Sent: 20 April 2010 11:45 To: Gupta,R,Ritesh,DKH C Cc: s...@hirschbichler.biz; sipp-users@lists.sourceforge.net Subject: Re: [Sipp-users] crazy problem on simple call scenario Me too getting same error in the error log files even though I'm just running sipp as a client and connecting to the actual voice mail server. :( :(. Please anyone can tell why its happening.?? Cheers, Rawat On Tue, Apr 20, 2010 at 15:07, <ritesh.gu...@bt.com<mailto:ritesh.gu...@bt.com>> wrote: Hi Michael, Can you please let me know how to split two UAC and UAS? Do we need to run two separate SIPp instance one for Register and one for Invite? In that case how they are going to map because Registration is done for particular number so how Invite instance going to understand that it should receive Invite for particular number? I tried two split UAC and UAS.. I run two separate sipp instance. "Instance A" for Register and "Instance B" for Invite. In that case also I am receiving same error on "Instance A" ---Discarding message which can't be mapped to a known.. Any suggestion any idea any help ? Thanks for support... Please find my XML for "Instance A" and "Instance B" "Instance A"---- XML <scenario name="Basic Sipstone UAC"> <!-- In client mode (sipp placing calls), the Call-ID MUST be --> <!-- generated by sipp. To do so, use [call_id] keyword. --> <send > <![CDATA[ REGISTER sip:10.230.53.225 SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: <sip:4...@10.230.53.227:5060<http://sip:4...@10.230.53.227:5060>> To: "420"<sip:4...@10.230.53.225<mailto:sip%3a...@10.230.53.225>> From: "420"<sip:4...@10.230.53.225<mailto:sip%3a...@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:4...@10.230.53.225<mailto:sip%3a...@10.230.53.225> SIP/2.0 Via: SIP/2.0/UDP 10.230.53.227:5060;branch=[branch];rport Max-Forwards: 70 Contact: <sip:4...@10.230.53.227:5060<http://sip:4...@10.230.53.227:5060>> To: "420"<sip:4...@10.230.53.225<mailto:sip%3a...@10.230.53.225>> From: "420"<sip:4...@10.230.53.225<mailto:sip%3a...@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 User-Agent: X-Lite release 4204o stamp 56125 Event: message-summary Content-Length: [len] ]]> </send> <recv response="501" crlf="true"> </recv> "Instance B" xml sample <scenario name="Basic Sipstone UAS"> <!-- In client mode (sipp placing calls), the Call-ID MUST be --> <!-- generated by sipp. To do so, use [call_id] keyword. --> <recv request="INVITE"> </recv> <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> -----Original Message----- From: Michael Hirschbichler [mailto:s...@hirschbichler.biz<mailto:s...@hirschbichler.biz>] Sent: 12 April 2010 07:35 To: sipp-users@lists.sourceforge.net<mailto:sipp-users@lists.sourceforge.net> Subject: Re: [Sipp-users] crazy problem on simple call scenario This scenario as described below won't work. If I understood the description correctly, the signalling-flow is UA Proxy ---REGISTER--> <---401------- ---REGISTER--> <---200------- <--INVITE----- .... In sipp, the mapping of a message (request/reply) is done by parsing for the SIP Call-ID - if a message is incoming with another call-id than the call-id in the originating request, the message is dropped as an unexpected message. In general, one sipp instance is not able to act as a UAC (for the registration process) and as an UAS (for the incomming invite request) at the same time. You have to split up the functionality to two sequenced sipp-instances: UA_C_ Proxy ---REGISTER--> <---401------- ---REGISTER--> <---200------- and after that UA_S_ Proxy <--INVITE----- ---180--------> ---200--------> .... hth and br Michael On 2010-04-09 17:12, Ruhi Aslan wrote: > ------------------------------------------------------------------------ > *De :* Ruhi Aslan > *Envoyé :* vendredi, 9. avril 2010 16:56 > *À :* > 'sipp-users-requ...@lists.sourceforge.net<mailto: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<http://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<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<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<http://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<http://asterisk.ch>;branch=z9hG4b-ID;rport=5060 > From: "43" <sip:4...@voip.vtx.ch<mailto:sip%3...@voip.vtx.ch>>;tag=as1cf8af76 > To: <sip:4...@mycomputerip:5060> > Contact: <sip:4...@_asterisk.ch_> > Call-ID: call...@asterisk.ch<mailto:call...@asterisk.ch> > <mailto:call...@asterisk.ch<mailto: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 > > ------------------------------------------------------------------------------ Download Intel® Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev _______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net<mailto:Sipp-users@lists.sourceforge.net> https://lists.sourceforge.net/lists/listinfo/sipp-users ------------------------------------------------------------------------------ Download Intel® Parallel Studio Eval Try the new software tools for yourself. Speed compiling, find bugs proactively, and fine-tune applications for parallel performance. See why Intel Parallel Studio got high marks during beta. http://p.sf.net/sfu/intel-sw-dev _______________________________________________ Sipp-users mailing list Sipp-users@lists.sourceforge.net<mailto:Sipp-users@lists.sourceforge.net> https://lists.sourceforge.net/lists/listinfo/sipp-users
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