[asterisk-users] test call generator

Daniel - Asterisk earohuanca at gmail.com
Tue Jun 28 14:55:19 CDT 2011


Hi List,

I'm trying to get working SIPp with media but something is wrong (it's
working well without media), please help:

This is the command I send at SIPp server:
      ./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err

This is the result I see:
      Last Error: Aborting call on unexpected message for Call-Id
'19-12768 at 12...

What I see at logs:

2011-06-28      14:32:57:624    1309289577.624809: Aborting call on
unexpected message for Call-Id '1-12768 at 127.0.0.1': while expecting '100'
(index 1), received 'SIP/2.0 488 Not acceptable here^M
Via: SIP/2.0/UDP 127.0.0.1:5061
;branch=z9hG4bK-12768-1-0;received=192.168.25.253^M
From: sipp <sip:sipp at 127.0.0.1:5061>;tag=12768SIPpTag091^M
To: sut <sip:2005 at 192.168.1.18:5060>;tag=as3614adc3^M
Call-ID: 1-12768 at 127.0.0.1^M
CSeq: 1 INVITE^M
Server: Asterisk PBX 1.8.4.1^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH^M
Supported: replaces, timer^M
Content-Length: 0^M

This is my asterisk 1.8's configuration:
*sip.conf*
[sipp]
type=friend
context=sipp
host=dynamic
port=6000
user=sipp
canreinvite=no
disallow=all
allow=ulaw
*
*
*extensions.conf:*
[sipp]
exten => 2005,1,Answer
same=>n,Dial(SIP/intern,30)
same=>n,Hangup()

exten => 2006,1,Answer()
same=> n,WaitMusicOnHold(4)
same=> n,Hangup()


I'm using sipp.3.1.src.tar.gz and I have installed it this way:
..sip.svn# make pcapplay

Thanks in advance.

Elder
On Thu, May 12, 2011 at 2:51 PM, Steve Totaro
<stotaro at asteriskhelpdesk.com>wrote:

> http://tinyurl.com/3hx5652
>
> On Thu, May 12, 2011 at 11:52 AM, Daniel - Asterisk <earohuanca at gmail.com>wrote:
>
>> Hello Everyone,
>>
>> I wonder if someone could share a manual about using SIPp for Asterisk's
>> testing.
>>
>> I'll be gratefull
>>
>>
>> Regards,
>>
>> Elder Arohuanca
>> Lima - Peru
>>
>>
>> On Tue, Sep 30, 2008 at 12:59 PM, zac wolfe <zac.wolfe at gmail.com> wrote:
>>
>>> Sipp looks pretty good! I don't know how I missed this one.  This
>>> would've saved me tons of time a couple months ago.
>>>
>>> I plan on using it to load test using 2 Asterisk servers, one to initiate
>>> the SIP calls, the other to receive. Thanks for the tip Alex.
>>>
>>> Zac Wolfe
>>> Safi Systems LLC
>>> www.safisystems.com
>>>
>>>
>>> On Sat, Sep 27, 2008 at 5:58 AM, Alex Balashov <
>>> abalashov at evaristesys.com> wrote:
>>>
>>>> What you are looking for is SIPP:   http://sipp.sourceforge.net/
>>>>
>>>> It won't intrinsically tell you anything about the data;  it's up to you
>>>> to appropriate the findings.  But it accomplishes the generation of
>>>> traffic (and dummy media!) on a technical level.
>>>>
>>>> Igor Hernandez wrote:
>>>>
>>>> > Sam Tam wrote:
>>>> >> Hello everyone
>>>> >>
>>>> >>
>>>> >>
>>>> >> I am trying to look for a free test call generator that will get me
>>>> some
>>>> >> stats like PDD, ASR and call quality etc on each route. As well as do
>>>> >> test at every interval too
>>>> >>
>>>> >>
>>>> >> If you know something like this please enlighten me.
>>>> >>
>>>> >> Sam
>>>> >>
>>>> >>
>>>> >>
>>>> ------------------------------------------------------------------------
>>>> >>
>>>> >> _______________________________________________
>>>> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>> >>
>>>> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>>>> >> Register Now: http://www.astricon.net
>>>> >>
>>>> >> asterisk-users mailing list
>>>> >> To UNSUBSCRIBE or update options visit:
>>>> >>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>> >
>>>> > Hey Sam,
>>>> >
>>>> > I've been looking for such a tool also. I can't seem to find a tool
>>>> that
>>>> > does those things.
>>>> >
>>>> > If nothing comes up in the next couple of weeks I'm going to code
>>>> > something up, I wouldn't mind letting you and anyone else who might be
>>>> > interested have the source once its done.
>>>> >
>>>> > Let me know if you find anything thats already out there in the
>>>> > meantime, might just save me a few hours of work.
>>>> >
>>>> > Regards,
>>>> >
>>>> >
>>>>
>>>>
>>>> --
>>>> Alex Balashov
>>>> Evariste Systems
>>>> Web    : http://www.evaristesys.com/
>>>> Tel    : (+1) (678) 954-0670
>>>> Direct : (+1) (678) 954-0671
>>>> Mobile : (+1) (706) 338-8599
>>>>
>>>> _______________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>>
>>>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>>>> Register Now: http://www.astricon.net
>>>>
>>>> asterisk-users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>>> Register Now: http://www.astricon.net
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _____________________________________________________________________
>>
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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