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
>
------------------------------------------------------------------------------
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/
_______________________________________________
Sipp-users mailing list
Sipp-users@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/sipp-users
