Hi Rob,
Thanks for the response. The IVR sends random INVITE's so I am attempting to
use the gettimeofday variable as you suggested.
I could not find many examples, but I think I am getting closer. For some
reason the XML script is not looping back to the BYE section.
This is what I see in the log;
elapsed_time=213000.000000 max_duration=120000.000000
elapsed_time=213000.000000 max_duration=120000.000000
When elapsed time is greater than or equal to max_duration it should send the
BYE but does not. Here is the relevant sections of code;
<nop>
<action>
<gettimeofday assign_to="start_time,t2"/>
<assign assign_to="duration" value="120000"/>
<multiply assign_to="start_time" value="1000"/>
</action>
</nop>
<label id="1" />
<nop>
<action>
<gettimeofday assign_to="t3,t4"/>
<assign assign_to="t2" variable="t4"/>
<multiply assign_to="t3" value="1000"/>
<subtract assign_to="t3" variable="start_time"/>
</action>
</nop>
<nop>
<action>
<test assign_to="9" variable="t3" compare="greater_than_equal"
value="120000" />
<log message="elapsed_time=[$t3] max_duration=[$duration] " />
</action>
</nop>
<recv request="INVITE" />
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
]]>
</send>
<recv request="ACK" next="1" />
<label id="9" />
<send retrans="500" test="9" next="1" >
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp
<sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
From: robertk...@gmail.com [mailto:robertk...@gmail.com] On Behalf Of Rob Day
Sent: Tuesday, 25 March 2014 6:48 AM
To: Brett Harding (AU)
Cc: sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] Cannot establish more than one simultaneous call
Brett,
Do you know how often the IVR is sending these INVITEs? If they're
regular (e.g. one every five minutes), it may make sense to just write
the XML script to expect six of these INVITEs and then send the BYE.
Otherwise, you could wait for an INVITE, then use the gettimeofday
action (http://sipp.sourceforge.net/doc/reference.html#gettimeofday)
to see if 30 minutes have passed - if they have, then send a BYE,
otherwise loop back and waiuut for another INVITE.
Best,
Rob
On 24 March 2014 04:52, Brett Harding (AU)
<brett.hard...@dimensiondata.com<mailto:brett.hard...@dimensiondata.com>> wrote:
> Hi Rob,
>
>
>
> Perhaps I was too hasty in my response. I can establish multiple,
> simultaneous calls, however the calls don't stay connected for the full
> duration of the final pause statement. From what I can tell, the IVR starts
> returning SIP INVITE messages after a few minutes into the call. I suspect
> these messages are acting as SIP keepalive messages? How would I keep sipp
> connected to the IVR for the full duration of the call (30 minutes) using a
> pause statement whilst also handling the inbound SIP keepalives?
>
>
>
> I have attached a copy of the XML scenario being used and the error /
> message log.
>
>
>
> This is the command I used to initiate the call.
>
> /usr/local/bin/sipp -i 10.ss.ss.ss -s 60661 10.cc.cc.cc -l 5 -r 2 -rp 2s -m
> 5 -sf webex_test.xml
>
>
>
> BR
>
> Brett
>
> From: Brett Harding (AU)
> Sent: Monday, 24 March 2014 1:42 PM
> To: 'Rob Day'
> Cc: sipp-users@lists.sourceforge.net<mailto:sipp-users@lists.sourceforge.net>
> Subject: RE: [Sipp-users] Cannot establish more than one simultaneous call
>
>
>
> Hi Rob,
>
>
>
> Thanks for responding. We recently upgraded our IVR server and so far the
> issue does not appear to be reoccurring.
>
>
>
> Brett
>
>
>
> From: robertk...@gmail.com<mailto:robertk...@gmail.com>
> [mailto:robertk...@gmail.com] On Behalf Of Rob
> Day
> Sent: Saturday, 22 March 2014 11:36 AM
> To: Brett Harding (AU)
> Cc: sipp-users@lists.sourceforge.net<mailto:sipp-users@lists.sourceforge.net>
> Subject: Re: [Sipp-users] Cannot establish more than one simultaneous call
>
>
>
>
>
> Is it possible for you to run the test again with -trace_msg and
> -trace_err? That should produce log files that give more detailed
> information on what's going wrong (for example, any Call-IDs being
> reused).
>
> Thanks,
> Rob
>
> On 18 March 2014 07:26, Brett Harding (AU)
> <brett.hard...@dimensiondata.com<mailto:brett.hard...@dimensiondata.com>>
> wrote:
>> Hello,
>>
>>
>>
>> I have an XML scenario which works perfectly for establishing a single
>> phone
>> call through an IVR system. When attempting to establish two or more
>> simultaneous calls, the scenario eventually fails with a "dead call msg"
>> error. Is it possible that I cannot re-use the same media ports in the XML
>> scenario file? What else am I missing?
>>
>>
>>
>> The command I use to generate the call is "sipp -i 10.x.x.x -s 4000
>> 10.x.x.x
>> -l 2 -r 2 -rp 2s -m 2 -sf webex.xml"
>>
>>
>>
>> I have attached the current XML scenario, if someone could take a peek
>> that
>> would be appreciated.
>>
>>
>>
>> Rgds
>>
>> Brett
>>
>>
>>
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>
>
> itevomcid
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