Also,
Please note that my scenario has to do with a caller calling a callee VIA a
PBX. Not directly (i.e., not sipp<->sipp). I frankly don't see where the
applicability of a direct calling scenario is.
Thanks,
Sonny.
On Wed, Mar 23, 2016 at 2:48 PM, Tickling Contest <
tickling.cont...@gmail.com> wrote:
> Tyler,
>
> Thanks, I am happy there is somebody else who shares my frustration with
> sipp (and I enjoyed the adjectives you used to describe the life of a
> SIPper on the github page). SIPp is very useful software, for sure, but it
> is godawfully badly documented (and uses (ugh!) XML) and there are many
> confusing handles. They will save a world of questions and trouble if they
> could just present an integrated scenario for a simple caller->callee
> scenario with auth using known SIP controllers like Asterisk.
>
> My hope is that if I ever get out of this process alive, I am going to
> document the crap out of this, and contribute it back here. I have consumed
> enough of everyone's time to warrant that sort of PSA.
>
> Anyway, how would you write the following scenario with pysipp?
>
> Callers (for a bunch of callers in CallerCSV.csv):
> ======================================
> REGISTER
> Send out INVITE
>
> (when INVITE was accepted)
> pause for a call duration selected from a uniform (10 s to 180 s)
> Send BYE
>
> Callees (for a bunch of callees in CalleeCSV.csv):
> ======================================
> REGISTER
> Wait for INVITE
> Accept the call
> Wait for BYE
> send 200 OK
>
> I am sold on pysipp if it can do this in the simplest possible fashion.
> The current example on your github page probably does this, but I wanted to
> check if there is something else I need to do.
>
> What would my caller.xml and callee.xml (any other xml needed?) look like?
> What are the referee and referer xmls in your page?
>
> Thanks!
>
>
> On Wed, Mar 23, 2016 at 2:47 PM, Tickling Contest <
> tickling.cont...@gmail.com> wrote:
>
>> Sorry, I did not send it to the correct mailing list originally:
>>
>> Tyler,
>>
>> Thanks, I am happy there is somebody else who shares my frustration with
>> sipp (and I enjoyed the adjectives you used to describe the life of a
>> SIPper on the github page). SIPp is very useful software, for sure, but it
>> is godawfully badly documented (and uses (ugh!) XML) and there are many
>> confusing handles. They will save a world of questions and trouble if they
>> could just present an integrated scenario for a simple caller->callee
>> scenario with auth using known SIP controllers like Asterisk.
>>
>> My hope is that if I ever get out of this process alive, I am going to
>> document the crap out of this, and contribute it back here. I have consumed
>> enough of everyone's time to warrant that sort of PSA.
>>
>> Anyway, how would you write the following scenario with pysipp?
>>
>> Callers (for a bunch of callers in CallerCSV.csv):
>> ======================================
>> REGISTER
>> Send out INVITE
>>
>> (when INVITE was accepted)
>> pause for a call duration selected from a uniform (10 s to 180 s)
>> Send BYE
>>
>> Callees (for a bunch of callees in CalleeCSV.csv):
>> ======================================
>> REGISTER
>> Wait for INVITE
>> Accept the call
>> Wait for BYE
>> send 200 OK
>>
>> I am sold on pysipp if it can do this in the simplest possible fashion.
>> The current example on your github page probably does this, but I wanted to
>> check if there is something else I need to do.
>>
>> What would my caller.xml and callee.xml (any other xml needed?) look
>> like? What are the referee and referer xmls in your page?
>>
>> Thanks!
>>
>> On Wed, Mar 23, 2016 at 12:04 PM, Tyler Goodlet <tgood...@gmail.com>
>> wrote:
>>
>>> Hey Tickling,
>>>
>>> Usually when you see a TCP connection error like that it means you're
>>> trying to contact a socket that isn't being listenend on.
>>> Just breifly looking over your shell scripts I can see that caller.sh
>>> isn't referencing the callee.sh $SIPPort anywhere.
>>> Are you routing to that port in the asterisk dialplan?
>>> So it might be simple issue that when asterisk bridges the two legs it's
>>> making the initial request to port 5060 on the b-leg and you don't have the
>>> callee SIPp agent listening there.
>>>
>>> On another note, (and I think I've recommended this to you before) you
>>> might want to try out pysipp https://github.com/SIPp/pysipp.
>>> It's going to let you avoid all these separate shell scripts and will
>>> make running the registration in sequence with the call a lot easier.
>>> If you want example code I'd be glad to modify your gist.
>>>
>>> Let me know!
>>>
>>>
>>> Tyler Goodlet
>>>
>>>
>>> On Mon, Mar 21, 2016 at 9:53 PM, Tickling Contest <
>>> tickling.cont...@gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> I am using an Asterisk PBX 13.7.2 which I would like to load test. For
>>>> this question, please use my gist at
>>>> https://gist.github.com/ticklingcontest/a67a4386cf73398efb82
>>>>
>>>> I am trying to load the PBX using calls generated by sipp using the
>>>> shell scripts in the gist in the following manner:
>>>>
>>>> (a) callee.sh is run first with the calleeCSVFile (with the fields
>>>> flipped; I do not show this CSV file in the gist)
>>>> (b) caller.sh is run next
>>>>
>>>> The hope is that callers are able to call the callee.
>>>>
>>>> Unfortunately, I see that the INVITEs are being received at the
>>>> Asterisk PBX, but they are never sent to the callee endpoints because:
>>>>
>>>> [Mar 21 21:24:56] ERROR[1535]: pjsip:0 <?>: tcpc0x7efd6807 TCP
>>>> connect() error: Connection refused [code=120111]
>>>> [Mar 21 21:24:56] WARNING[1535]: pjsip:0 <?>: tsx0x7efd6808d Failed to
>>>> send Request msg INVITE/cseq=17210 (tdta0x7efd68085a80)! err=120111
>>>> (Connection refused)
>>>>
>>>> How do I fix this?
>>>>
>>>> Any help is appreciated.
>>>>
>>>> Also, can I please get some help with my *.xml files? All my
>>>> registrations end up getting the following error even though I use the -aa
>>>> parameter:
>>>> Aborted call with Call-ID 'de92a989-dea1-45a3-835f-07d57fcddd76'. (The
>>>> NOTIFY that is received has the same call-id).
>>>>
>>>> BTW, my caller script works when calling a real SIP endpoint (on my
>>>> phone). I just cannot get it to work on this script.
>>>>
>>>> Can someone tell me how I can make a simple call using endpoints that
>>>> connect via Asterisk AFTER two endpoints REGISTER at a PBX? Do I need 3 XML
>>>> files like I use (1 for registration and 1 each for caller and callee) or
>>>> is there a better way? If I do need separate XML files, how do I make sure
>>>> that a number of endpoints on a sipp instance are found by the call
>>>> receiving instance of sipp?
>>>>
>>>> Any help is appreciated.
>>>>
>>>> Thanks.
>>>>
>>>>
>>>> ------------------------------------------------------------------------------
>>>> Transform Data into Opportunity.
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>>>>
>>>>
>>>
>>>
>>> ------------------------------------------------------------------------------
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>>>
>>>
>>
>
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