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
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