Hello,

I am trying to run load on asterisk server(version 1.8.7.1) for the voicemail() application using SIPp tool. I am just running sipp at call rate of 1 cps with the following command:

./sipp -m 9000 -r 1 -rp 1000 -d 45 -max_socket 65536 -sf uac_msg_deposit.xml -i 172.16.129.13 -s 1 172.16.129.14 –trace_err

I am trying to deposit 9000 messages in the mailbox of user 1 (given by the -s option) but the following  warning is coming on the asterisk server due to which the message does not get deposited into the users mailbox:

No audio available on SIP/172.16.129.13:5060-00000001??

I have set rtpstart=6000 and rtpend=20000 in rtp.conf.

Can someone please let me know how to avoid these kind of warnings.

Thanks.

Shalu

Thanks and Regards,
Shalu Dhamija
Rancore Technologies(P) Ltd.
Gurgaon
Ph : 0124-4200691
+91-9910995356(M)

Jared,

Thank you for that information!

Has anyone else had an experience like this?

On 12 May 2011 20:25, Jared Geiger wrote:

> Hi David,
>
> When I was testing 1.6.1 for high volume channels, I couldn’t get over 1000
> channels / 40 CPS without the load average spiking up due to io wait. I
> switched back to 1.4 and I can go to 3000 channels / 75 CPS with no io wait
> and a load average in the 1s. It seemed like it was caused by the new timing
> system in 1.6.1 even though I wasn’t proxying media using only SIP.
>
> I haven’t tried 1.8 yet to see if it handles large call volumes any better.
>
> ~Jared
>
> On Wed, May 11, 2011 at 8:29 PM, David Cunningham <
> dcunningham@voisonics.com> wrote:
>
>> Hello,
>>
>> We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now
>> experiencing higher CPU utilization on their server. I can’t see anything
>> wrong, so is this just expected with 1.6? Can anyone help explain it?
>>
>> Thanks for any advice.
>>
>> –
>> David Cunningham, Voisonics
>> http://voisonics.com/
>> US toll-free: +1 888 842 2720
>> UK: +44 (0) 20 3298 1642
>> Australia: +61 (0) 2 8063 9019
>>
>>
>> –
>> _____________________________________________________________________
>> — 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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>> To UNSUBSCRIBE or update options visit:
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>>
>
>
> –
> _____________________________________________________________________
> — 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>

Hi
Asterisk re-started automatically with 10 CPS(each call duration was 5sec)
Has someone met something similar ?
I could see in the log that asterisk segmentation fault and restart.

Is there any idea how to regulate attack of invite to prevent restart?

version 1.6.1.11
CentOS 5.3
Astdatabase 5000 lines (using database 1 line every 1call)
cpu and memory usage is moderate(cpu MHz :1800.000 MemoryTotal:2075456 kB)

thanks

messages.zip