but how do i get the ACK and BYE signals? or i have to use something like
this <recv request="REGISTER|SUBSCRIBE" rrs="true"> or labels....?
On Wed, Jul 20, 2011 at 10:14 PM, Greg Henderson <gnhenderso...@gmail.com>wrote:
> Your scenario needs to expect the SUBSCRIBE and act accordingly to the
> expect SIP dialog(s) that will happen. this isn't an error in SIPp as much
> as in the scenario file.
> On Jul 20, 2011, at 11:33 AM, Gopal krishnan wrote:
>
> Hi,
>
> Try like this, actually I tried with your registration script, and I tried
> to register two extensions, it got registered, but from the log i saw
> something like the below,
> 2011-07-20 21:43:59:097 1311178439.097676: Aborting call on unexpected
> message for Call-Id '6219001052ac4097be2d4dd57450b8ad': while expecting
> 'REGISTER' (index 0), received 'SUBSCRIBE sip:2002@192.168.0.110 SIP/2.0
>
> Can you try the following, and see what happens
> 1. run ./sipp -i <local ip address> -p 5060 -sf reg.xml in one window
> 2. register two softphone with extension 2001 and 2002
> 3. run ./sipp -sf uas.xml in another window
> 4. from 2001 try calling 2001 or 2002
> 5. you will see some traffic in uas.xml screen
>
> even for me also I am not able to see the bye signal here.
>
> Any assistance would be appreciated.
>
>
> On Wed, Jul 20, 2011 at 5:45 PM, Bui Dinh Thang
> <buidinhthan...@gmail.com>wrote:
>
>> hj
>>
>>
>> 2011/7/20 Gopal krishnan <gopalakrishnan...@gmail.com>
>>
>>> I hope using with single xml file is good.
>>>
>>> And from your softphone how you are trying to establish the connection?
>>> by dialing any other softphones number? so that i will try the same setup in
>>> my place.
>>>
>>>
>>> On Wed, Jul 20, 2011 at 4:02 PM, Bui Dinh Thang <
>>> buidinhthan...@gmail.com> wrote:
>>>
>>>> hj!
>>>> i using a registration script for registering for my softphone.i have
>>>> used registration together with UAS script, but it didn't work, so i using
>>>> two script for that. is it ok?
>>>> regards
>>>>
>>>>
>>>> 2011/7/20 Gopal krishnan <gopalakrishnan...@gmail.com>
>>>>
>>>>> How you are registering your softphone to SIPp, are you using any
>>>>> registration script for that?
>>>>>
>>>>>
>>>>> On Wed, Jul 20, 2011 at 6:35 AM, Bui Dinh Thang <
>>>>> buidinhthan...@gmail.com> wrote:
>>>>>
>>>>>> hj!
>>>>>> i didn't use any IPPBX between softphone and sipp(UAS), i use one
>>>>>> softphone and one Sipp to test UAS of sipp.my script in below, when i try
>>>>>> run this script in my machine, sometime it's worked, but when i restart
>>>>>> the
>>>>>> server, it's didn't work, i don't know why???
>>>>>> maybe softphone is error??? but i try in SIpInspector it's ok,that's
>>>>>> crazy.
>>>>>> regards
>>>>>> thanks
>>>>>>
>>>>>> 2011/7/19 Gopal krishnan <gopalakrishnan...@gmail.com>
>>>>>>
>>>>>>> Can you let us know how you are trying to setup the UAS and your
>>>>>>> softphone; are you using any IPPBX inbetween softphone and SIPp (UAS).
>>>>>>> It
>>>>>>> would help us to test the same in my place.
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Jul 19, 2011 at 4:36 PM, Bui Dinh Thang <
>>>>>>> buidinhthan...@gmail.com> wrote:
>>>>>>>
>>>>>>>> hj all! can you help me!
>>>>>>>> i don't know why my softphone can't send any message to my sipp
>>>>>>>> server??? ACK, BYE. i use wireshark to check, but i didn't see any
>>>>>>>> message
>>>>>>>> from phone send to sipp.
>>>>>>>> regard
>>>>>>>> thanks
>>>>>>>>
>>>>>>>> 2011/7/19 Bui Dinh Thang <buidinhthan...@gmail.com>
>>>>>>>>
>>>>>>>>> Thanks reply!i will try your suggest
>>>>>>>>> regard
>>>>>>>>>
>>>>>>>>> 2011/7/19 Patrick Wakano <pwak...@gmail.com>
>>>>>>>>>
>>>>>>>>>> The first thing you have to fix is the usage of the record-route.
>>>>>>>>>> When you receive a Record-route field you are not supposed to
>>>>>>>>>> reinsert it in the response as you are doing with the field
>>>>>>>>>> "[last_Record-Route]". Basically, you should send all your responses
>>>>>>>>>> to the
>>>>>>>>>> servers listed in the received Record-Route field. In your SIPp
>>>>>>>>>> script you
>>>>>>>>>> can do it, by using the rrs=true in the received request and
>>>>>>>>>> inserting
>>>>>>>>>> the keyword [routes] in the response.
>>>>>>>>>> See if this solves your problem.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Mon, Jul 18, 2011 at 11:59 AM, Bui Dinh Thang <
>>>>>>>>>> buidinhthan...@gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hj! thanks your reply!
>>>>>>>>>>> But the script you sent, I've done but still having the same
>>>>>>>>>>> error, phone not sending ACK and BYE message to the sipp, i
>>>>>>>>>>> don't know what's problem happening????
>>>>>>>>>>> regard
>>>>>>>>>>>
>>>>>>>>>>> 2011/7/18 Gopal krishnan <gopalakrishnan...@gmail.com>
>>>>>>>>>>>
>>>>>>>>>>>> Try the attached script, this one from SIPp. If this script
>>>>>>>>>>>> works then include your pcap audio file.
>>>>>>>>>>>>
>>>>>>>>>>>> On Mon, Jul 18, 2011 at 4:48 PM, Bui Dinh Thang <
>>>>>>>>>>>> buidinhthan...@gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> hj all!
>>>>>>>>>>>>> i try to simulate the scenario, but i don't know what problem,
>>>>>>>>>>>>> phone can not send the ACK and BYE to sipp, i don't understand
>>>>>>>>>>>>> it.
>>>>>>>>>>>>>
>>>>>>>>>>>>> this my scenario!
>>>>>>>>>>>>>
>>>>>>>>>>>>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>>>>>>>>>>>>> <scenario name="UASBasic">
>>>>>>>>>>>>>
>>>>>>>>>>>>> <recv request="INVITE" crlf="true" rrs="true">
>>>>>>>>>>>>> </recv>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <send>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <![CDATA[
>>>>>>>>>>>>>
>>>>>>>>>>>>> SIP/2.0 100 Trying
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_To:];tag=[pid]SIPpTag01[call_number]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_From:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_Call-ID:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_CSeq:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> Contact: [field0] <sip:[local_ip]:[local_port]>
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_Via:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> User-Agent: uas
>>>>>>>>>>>>>
>>>>>>>>>>>>> Content-Length: 0
>>>>>>>>>>>>>
>>>>>>>>>>>>> ]]>
>>>>>>>>>>>>>
>>>>>>>>>>>>> </send>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <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]>
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_record-Router:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> Content-Length: 0
>>>>>>>>>>>>>
>>>>>>>>>>>>> ]]>
>>>>>>>>>>>>>
>>>>>>>>>>>>> </send>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <send>
>>>>>>>>>>>>> <![CDATA[
>>>>>>>>>>>>> SIP/2.0 183 Session Progress
>>>>>>>>>>>>> [last_Via:]
>>>>>>>>>>>>> [last_From:]
>>>>>>>>>>>>> [last_To:]
>>>>>>>>>>>>> [last_Call-ID:]
>>>>>>>>>>>>> [last_CSeq:]
>>>>>>>>>>>>> User-Agent: SIP
>>>>>>>>>>>>> Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>>>>>>>>>>>>> Content-Length: 0
>>>>>>>>>>>>> ]]>
>>>>>>>>>>>>> </send>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <send retrans="500">
>>>>>>>>>>>>>
>>>>>>>>>>>>> <![CDATA[
>>>>>>>>>>>>>
>>>>>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_Via:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_From:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_To:] ;tag=[call_number]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_Call-ID:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_CSeq:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_Record-Route:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> Supported: timer
>>>>>>>>>>>>>
>>>>>>>>>>>>> Contact: <sip:[local_ip]:[local_port];transport=[transport]>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Content-Type: application/sdp
>>>>>>>>>>>>>
>>>>>>>>>>>>> Content-Length: [len]
>>>>>>>>>>>>>
>>>>>>>>>>>>> v=0
>>>>>>>>>>>>>
>>>>>>>>>>>>> o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>>>>>>>>>>>>>
>>>>>>>>>>>>> s= SIPp - UAS
>>>>>>>>>>>>>
>>>>>>>>>>>>> c=IN IP[media_ip_type] [media_ip]
>>>>>>>>>>>>>
>>>>>>>>>>>>> t=0 0
>>>>>>>>>>>>>
>>>>>>>>>>>>> m=audio [media_port] RTP/AVP 0
>>>>>>>>>>>>>
>>>>>>>>>>>>> a=rtpmap:0 PCMU/8000
>>>>>>>>>>>>>
>>>>>>>>>>>>> a=sendrecv
>>>>>>>>>>>>>
>>>>>>>>>>>>> a=rtpmap:98 telephone-event/8000
>>>>>>>>>>>>>
>>>>>>>>>>>>> ]]>
>>>>>>>>>>>>>
>>>>>>>>>>>>> </send>
>>>>>>>>>>>>> *<recv request="ACK" rtd="true" crlf="true">*
>>>>>>>>>>>>>
>>>>>>>>>>>>> </recv>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <nop>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <action>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <exec play_pcap_audio="pcap/test.pcap"/>
>>>>>>>>>>>>>
>>>>>>>>>>>>> </action>
>>>>>>>>>>>>>
>>>>>>>>>>>>> </nop>
>>>>>>>>>>>>>
>>>>>>>>>>>>> *<recv request="BYE">*
>>>>>>>>>>>>>
>>>>>>>>>>>>> </recv>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <send>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <![CDATA[
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> SIP/2.0 200 OK
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_Via:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_From:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_To:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_Call-ID:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> [last_CSeq:]
>>>>>>>>>>>>>
>>>>>>>>>>>>> Contact: [field0] <
>>>>>>>>>>>>> sip:[local_ip]:[local_port];transport=[transport]>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Content-Length: 0
>>>>>>>>>>>>>
>>>>>>>>>>>>> ]]>
>>>>>>>>>>>>>
>>>>>>>>>>>>> </send>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <!-- definition of the response time repartition table (unit is
>>>>>>>>>>>>> ms) -->
>>>>>>>>>>>>>
>>>>>>>>>>>>> <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150,
>>>>>>>>>>>>> 200"/>
>>>>>>>>>>>>>
>>>>>>>>>>>>> <!-- definition of the call length repartition table (unit is
>>>>>>>>>>>>> ms) -->
>>>>>>>>>>>>>
>>>>>>>>>>>>> <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000,
>>>>>>>>>>>>> 10000"/>
>>>>>>>>>>>>>
>>>>>>>>>>>>> </scenario>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> please help me show the errors my scenario?
>>>>>>>>>>>>> Best Regard!
>>>>>>>>>>>>> thanks
>>>>>>>>>>>>> --
>>>>>>>>>>>>> Thắng
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> ------------------------------------------------------------------------------
>>>>>>>>>>>>> AppSumo Presents a FREE Video for the SourceForge Community by
>>>>>>>>>>>>> Eric
>>>>>>>>>>>>> Ries, the creator of the Lean Startup Methodology on "Lean
>>>>>>>>>>>>> Startup
>>>>>>>>>>>>> Secrets Revealed." This video shows you how to validate your
>>>>>>>>>>>>> ideas,
>>>>>>>>>>>>> optimize your ideas and identify your business strategy.
>>>>>>>>>>>>> http://p.sf.net/sfu/appsumosfdev2dev
>>>>>>>>>>>>> _______________________________________________
>>>>>>>>>>>>> Sipp-users mailing list
>>>>>>>>>>>>> Sipp-users@lists.sourceforge.net
>>>>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>> Thắng
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> ------------------------------------------------------------------------------
>>>>>>>>>>> AppSumo Presents a FREE Video for the SourceForge Community by
>>>>>>>>>>> Eric
>>>>>>>>>>> Ries, the creator of the Lean Startup Methodology on "Lean
>>>>>>>>>>> Startup
>>>>>>>>>>> Secrets Revealed." This video shows you how to validate your
>>>>>>>>>>> ideas,
>>>>>>>>>>> optimize your ideas and identify your business strategy.
>>>>>>>>>>> http://p.sf.net/sfu/appsumosfdev2dev
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Sipp-users mailing list
>>>>>>>>>>> Sipp-users@lists.sourceforge.net
>>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> Thắng
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Thắng
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Thắng
>>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Thắng
>>>>
>>>
>>>
>>
>>
>> --
>> Thắng
>>
>
>
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