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Re: [Sipp-users] Connecting phone system to SIPp
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From: Rob D. <rk...@rk...> - 2014-04-08 19:11:43
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Hi David, Do you know the SIP URIs you want to use? If so, you can simply create a custom scenario file (e.g. by running `sipp -sd uac > custom-uac.xml`), edit that custom scenario file to use the right SIP URIs in each message, then use that scenario rather than the built-in one (by using `-sf custom-uac.xml` instead of `-sn uac`). Best, Rob On 8 April 2014 19:44, Munzer,David J <mu...@uf...> wrote: > Dear Rob, > > The default scenario is making a call to the address/port of Kamailio. > How do I include both the address of the SIP server that I making a call > to and the address of one of the actual registered users? The > authorization page did not clarify that well. > > Thankfully, > David > > > -----Original Message----- > From: Rob Day <rk...@rk...> > To: Munzer,David J <mu...@uf...> > Cc: sipp-users <sip...@li...> > Sent: Tue, Apr 8, 2014 3:04 am > Subject: Re: [Sipp-users] Connecting phone system to SIPp > > 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 <mu...@uf...> 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 <rk...@rk...> >> To: davidjmunzer <dav...@ao...> >> Cc: sipp-users <sip...@li...> >> 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, <dav...@ao...> 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 <rk...@rk...> >>> To: davidjmunzer <dav...@ao...> >>> Cc: sipp-users <sip...@li...> >>> 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, <dav...@ao...> wrote: >>>> The SIP server that I am using is Kamailio. >>>> >>>> >>>> -----Original Message----- >>>> From: Rob Day <rk...@rk...> >>>> To: Munzer,David J <mu...@uf...> >>>> Cc: sipp-users <sip...@li...> >>>> 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 <mu...@uf...> 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 >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ------------------------------------------------------------------------------ >>>>> Learn Graph Databases - Download FREE O'Reilly Book >>>>> "Graph Databases" is the definitive new guide to graph databases >>>>> and >>>>> their >>>>> applications. Written by three acclaimed leaders in the field, >>>>> this first edition is now available. Download your free book >>>>> today! >>>>> http://p.sf.net/sfu/13534_NeoTech >>>>> _______________________________________________ >>>>> Sipp-users mailing list >>>>> Sip...@li... >>>>> https://lists.sourceforge.net/lists/listinfo/sipp-users >>>> >>>> >>>> >>>> >>>> >>>> ------------------------------------------------------------------------------ >>>> >>>> _______________________________________________ >>>> Sipp-users mailing list >>>> Sip...@li... >>>> https://lists.sourceforge.net/lists/listinfo/sipp-users >>>> >>> >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Sipp-users mailing list >>> Sip...@li... >>> https://lists.sourceforge.net/lists/listinfo/sipp-users >>> >> >> >> ------------------------------------------------------------------------------ >> Put Bad Developers to Shame >> Dominate Development with Jenkins Continuous Integration >> Continuously Automate Build, Test & Deployment >> Start a new project now. Try Jenkins in the cloud. >> http://p.sf.net/sfu/13600_Cloudbees >> _______________________________________________ >> Sipp-users mailing list >> Sip...@li... >> https://lists.sourceforge.net/lists/listinfo/sipp-users > > ------------------------------------------------------------------------------ > Put Bad Developers to Shame > Dominate Development with Jenkins Continuous Integration > Continuously Automate Build, Test & Deployment > Start a new project now. Try Jenkins in the cloud. > http://p.sf.net/sfu/13600_Cloudbees > _______________________________________________ > Sipp-users mailing list > Sip...@li... > https://lists.sourceforge.net/lists/listinfo/sipp-users |
