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:[email protected] 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 <[email protected]>wrote:
> hj
>
>
> 2011/7/20 Gopal krishnan <[email protected]>
>
>> 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 <[email protected]
>> > 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 <[email protected]>
>>>
>>>> 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 <
>>>> [email protected]> 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 <[email protected]>
>>>>>
>>>>>> 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 <
>>>>>> [email protected]> 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 <[email protected]>
>>>>>>>
>>>>>>>> Thanks reply!i will try your suggest
>>>>>>>> regard
>>>>>>>>
>>>>>>>> 2011/7/19 Patrick Wakano <[email protected]>
>>>>>>>>
>>>>>>>>> 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 <
>>>>>>>>> [email protected]> 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 <[email protected]>
>>>>>>>>>>
>>>>>>>>>>> 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 <
>>>>>>>>>>> [email protected]> 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
>>>>>>>>>>>> [email protected]
>>>>>>>>>>>> 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
>>>>>>>>>> [email protected]
>>>>>>>>>> https://lists.sourceforge.net/lists/listinfo/sipp-users
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Thắng
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Thắng
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Thắng
>>>>>
>>>>
>>>>
>>>
>>>
>>> --
>>> Thắng
>>>
>>
>>
>
>
> --
> Thắng
>
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