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:[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
>
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>
> ------------------------------------------------------------------------------
> 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.
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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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