David,

The default scenario makes a call to sip:service@10.0.0.160:5060, as
you can see in the error log - is this user configured and registered
in Kamailio? (The -s option can change the "service" part, e.g. to a
phone number of your choosing.)

It may be worth looking at what logs Kamailio has - those should help
determine why it returns a 404.

Best,
Rob

On 8 April 2014 03:44, Munzer,David J <mund...@ufl.edu> wrote:
> Hi Rob,
>
> I tried your suggestion of inputting the "./sipp -sn uac -i (My
> computer's IP address) (Kamailio's IP address) -trace_err". SIPP is
> recognizing the Kamailio server but it fails after sending the Invite
> 100 message. It gave this message, 2: Aborting call on unexpected
> message for Call-Id '1-3288@10.0.0.210': while ex  'ecting '100' (index
> 1), received 'SIP/2.0 404 Not Found. I've considered that i may need to
> run on a different integrated scenario and that the error could come
> from Kamailio not authorizing the call due to me not providing further
> account information. However, if it was this case, I believe I would
> have received a 401 (Unauthorized) or a 407 (Proxy Authentication
> Required). If you have any ideas, please let me know. Below is the
> attempted call.
>
> Thankfully, David Munzer
>
> $ ./sipp -sn uac -i (My computer's IP address) (Kamailio's IP address)
> -trace_err
> Warning: open file limit > FD_SETSIZE; limiting max. # of open files to
> FD_SETSI
>                                                ZE = 64
> Resolving remote host '10.0.0.160'... Done.
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
>    Call-rate(length)   Port   Total-time  Total-calls  Remote-host
>    10.0(0 ms)/1.000s   5060      10.60 s          106
> 10.0.0.160:5060(UDP)
>
>    0 new calls during 0.000 s period      0 ms scheduler resolution
>    0 calls (limit 30)                     Peak was 1 calls, after 0 s
>    0 Running, 109 Paused, 0 Woken up
>    0 dead call msg (discarded)            0 out-of-call msg (discarded)
>    1 open sockets
>
>                                   Messages  Retrans   Timeout
> Unexpected-Msg
>        INVITE ---------->         106       0         0
>           100 <----------         0         0         0         106
>           180 <----------         0         0         0         0
>           183 <----------         0         0         0         0
>           200 <----------  E-RTD1 0         0         0         0
>           ACK ---------->         0         0
>         Pause [      0ms]         0                             0
>           BYE ---------->         0         0         0
>           200 <----------         0         0         0         0
>
> ------------------------------ Test Terminated
> --------------------------------
>
>
> ----------------------------- Statistics Screen ------- [1-9]: Change
> Screen --
>    Start Time             | 2014-04-05   02:20:42:441
> 1396678842.441802
>    Last Reset Time        | 2014-04-05   02:20:53:081
> 1396678853.081802
>    Current Time           | 2014-04-05   02:20:53:082
> 1396678853.082802
> -------------------------+---------------------------+--------------------------
>    Counter Name           | Periodic value            | Cumulative value
> -------------------------+---------------------------+--------------------------
>    Elapsed Time           | 00:00:00:001              | 00:00:10:641
>    Call Rate              |    0.000 cps              |    9.961 cps
> -------------------------+---------------------------+--------------------------
>    Incoming call created  |        0                  |        0
>    OutGoing call created  |        0                  |      106
>    Total Call created     |                           |      106
>    Current Call           |        0                  |
> -------------------------+---------------------------+--------------------------
>    Successful call        |        0                  |        0
>    Failed call            |        0                  |      106
> -------------------------+---------------------------+--------------------------
>    Response Time 1        | 00:00:00:000              | 00:00:00:000
>    Call Length            | 00:00:00:000              | 00:00:00:004
> ------------------------------ Test Terminated
> --------------------------------
>
> 2014-04-05      02:20:53:077    1396678853.077802: Aborting call on
> unexpected m
>                                                    essage for Call-Id
> '106-3288@10.0.0.210': while expecting '100' (index 1), recei
>
>                            ved 'SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 10.0.0.210:5060;branch=z9hG4bK-3288-106-0;rport=5060
>  From: sipp <sip:sipp@10.0.0.210:5060>;tag=3288SIPpTag00106
> To: sut
> <sip:service@10.0.0.160:5060>;tag=fc4b70b0517cb156b1fb39a76698f743-5763
> Call-ID: 106-3288@10.0.0.210
> CSeq: 1 INVITE
> Server: kamailio (4.0.4 (i386/linux))
> Content-Length: 0
>
> '.
> sipp: There were more errors, see 'uac_3288_errors.log' file
>
>
>
>
> -----Original Message-----
>  From: Rob Day <r...@rkd.me.uk>
> To: davidjmunzer <davidjmun...@aol.com>
> Cc: sipp-users <sipp-users@lists.sourceforge.net>
> Sent: Wed, Apr 2, 2014 2:24 pm
> Subject: Re: [Sipp-users] Connecting phone system to SIPp
>
> David,
>
> I think that is the wrong syntax - if you have Kamailio's IP address
> after the -i, SIPp will try to bind that IP address (which it doesn't
> own) and fail. You need "./sipp -sn uac -i <your local network IP
> address> <Kamailio machine IP address>"
>
> Note that you need to use the IP address that can talk to your network
> (the one which Kamailio is on, probably 192.168.x.x), not 127.0.0.1
> (which is localhost-only).
>
> Best,
> Rob
>
> On 2 April 2014 20:22,  <davidjmun...@aol.com> wrote:
>> Hi Rob,
>>
>> I tried doing that by imputing ./sipp -sn uac 127.0.0.1 -i IP dress,
>> It
>> responds with the error message, 1396509142.105827: Unable to bind
>> main
>> socket, errno = 125 (Cannot assign requested address). Is there an
>> issue
>> with my syntax, since I don't see why SIPP shouldn't be able to
>> access
>> Kamailio's IP address.
>>
>> Thankfully
>> David
>>
>>
>>
>>
>> -----Original Message-----
>> From: Rob Day <r...@rkd.me.uk>
>> To: davidjmunzer <davidjmun...@aol.com>
>> Cc: sipp-users <sipp-users@lists.sourceforge.net>
>> Sent: Wed, Apr 2, 2014 12:54 pm
>> Subject: Re: [Sipp-users] Connecting phone system to SIPp
>>
>> Hi David,
>>
>> I think this may be because your Windows machine provides its IPv6 or
>> its localhost address first, so SIPp uses that and is then unable to
>> send messages to other IPv4 machines on the network. If you
>> explicitly
>> specify an IP address to bind to (with the -i option) you should get
>> better results.
>>
>> Best,
>> Rob
>>
>> On 2 April 2014 19:18,  <davidjmun...@aol.com> wrote:
>>> The SIP server that I am using is Kamailio.
>>>
>>>
>>> -----Original Message-----
>>> From: Rob Day <r...@rkd.me.uk>
>>> To: Munzer,David J <mund...@ufl.edu>
>>> Cc: sipp-users <sipp-users@lists.sourceforge.net>
>>> Sent: Wed, Mar 26, 2014 1:14 pm
>>> Subject: Re: [Sipp-users] Connecting phone system to SIPp
>>>
>>> Rob,
>>>
>>> By phone system, I do mean SIP server, specifically a combination of
>>> Kamailio and Freeswitch. When I try running the program using
>>> "./sipp -sn
>>> uac the ip address",  It informs me  that it's unable to send UDP
>>> message:
>>> Bad address. I've checked that the SIP server's address is correct
>>> by
>>> doing
>>> ip add on the SIP server to verify the IP address.  Any ideas how to
>>> approach this issue?
>>>
>>> Thankfully,
>>> David
>>>
>>> David,
>>>
>>> When you say that you have a phone system running, do you mean that
>>> you have a SIP server (Kamailio/Clearwater/Asterisk) running, or
>>> something else?
>>>
>>> If you have a SIP server, it is probably listening on port 5060
>>> (though you can check by running `netstat -lnp`) and you can just
>>> give
>>> the IP address of that machine as a command-line argument to SIPp.
>>> I'm
>>> assuming you want to use SIPp in UAC mode to test this phone system
>>> -
>>> if you want SIPp in UAS mode, handling calls sent to it by that SIP
>>> server, you'll need to check the documentation of that SIP server to
>>> find how to configure it.
>>>
>>> SIPp only communicates through the SIP protocol, so if by 'phone
>>> system' you don't mean a SIP server, you'll have to set one up to
>>> translate between SIP and whatever phone system you have.
>>>
>>> Best,
>>> Rob
>>>
>>> On 25 March 2014 19:14, Munzer,David J <mund...@ufl.edu> wrote:
>>>> Hi,
>>>>
>>>> I have just finished installing SIPp and am not sure how to connect
>>>> my
>>>> phone system to SIPp. My computer is running the program on Windows
>>>> 7
>>>> through Cygwin. My phone system runs on Alpine Linuz through a USB.
>>>> Because of the two different operating systems, I need need to
>>>> connect
>>>> the two most likely through the IP address. However, I am unsure
>>>> how to
>>>> go about this. I would really appreciate help on this matter.
>>>>
>>>> Thankfully,
>>>> David
>>>>
>>>>
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
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>>>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------------
>>>
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>>> Sipp-users@lists.sourceforge.net
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>>
>>
>>
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