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
------------------------------------------------------------------------------
10 Tips for Better Web Security
Learn 10 ways to better secure your business today. Topics covered include:
Web security, SSL, hacker attacks & Denial of Service (DoS), private keys,
security Microsoft Exchange, secure Instant Messaging, and much more.
http://www.accelacomm.com/jaw/sfnl/114/51426210/
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