<div dir="ltr">Hi Shalu,<br><br>How you are invoking call in dialplan. it's completely depends on that. <br>And error show that no voice is there for store in voicemail .<br><br><div class="gmail_quote">On Wed, Jan 11, 2012 at 10:05 AM, shalu dhamija <span dir="ltr"><<a href="mailto:shalu.dhamija@rancoretech.com">shalu.dhamija@rancoretech.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><p>Hello,</p>
<p> </p>
<p>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:</p>
<p> </p>
<p>./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</p>
<p> </p>
<p>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:</p>
<p> </p>
<p><font color="#000099">No audio available on SIP/172.16.129.13:5060-00000001??</font></p>
<p><font color="#000099"></font> </p>
<p><font color="#000000">I have set rtpstart=6000 and rtpend=20000 in rtp.conf.</font></p>
<p> </p>
<p> </p>
<p>Can someone please let me know how to avoid these kind of warnings.</p>
<p> </p>
<p>Thanks.</p>
<p> </p>
<p>Shalu</p>
<p> </p>
<p> </p>
<p><br><br>Thanks and Regards,<br><font color="#888888">Shalu Dhamija<br>Rancore Technologies(P) Ltd.<br>Gurgaon<br>Ph : <a href="tel:0124-4200691" value="+911244200691" target="_blank">0124-4200691</a><br><a href="tel:%2B91-9910995356" value="+919910995356" target="_blank">+91-9910995356</a>(M)<br>
</font></p></div><br>--<br>
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