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> 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>
> To: "420"<sip:4...@10.230.53.225 <sip%3a...@10.230.53.225>>
> From: "420"<sip:4...@10.230.53.225 <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 <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>
> To: "420"<sip:4...@10.230.53.225 <sip%3a...@10.230.53.225>>
> From: "420"<sip:4...@10.230.53.225 <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]
> Sent: 12 April 2010 07:35
> To: 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'
> > *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 <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>
> > 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
> >
> >
>
>
>
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>
> ------------------------------------------------------------------------------
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