<div dir="ltr">Kristian:<div><br></div><div>Thanks for your reply. I am running asterisk as root, but still getting this error.</div><div><br></div><div>I did a test while running asterisk 1.4.21 version setting "ulimit -n 32768", but after restaring asterisk it stop working with less than 150 calls (less than 300 channels).</div>
<div><br></div><div>Any suggestion??</div><div><br></div><div><br><div class="gmail_quote">On Fri, Oct 10, 2008 at 11:37 AM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kkielhofner@star2star.com">kkielhofner@star2star.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div><div></div><div class="Wj3C7c">On 10/10/08, Juan Rodríguez <<a href="mailto:jerdguez@gmail.com">jerdguez@gmail.com</a>> wrote:<br>
> After getting some ERRORS like this:<br>
><br>
> [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup<br>
> media stream for this call.<br>
> [Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup<br>
> media stream for this call.<br>
> [Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup<br>
> media stream for this call.<br>
> [Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup<br>
> media stream for this call.<br>
><br>
> I start getting:<br>
><br>
> ERROR[14844] chan_sip.c: Unable to build sip pvt data for<br>
> 'TRUNK/DESTINATION' (Out of memory or socket error)<br>
> [Oct 9 22:26:45] ERROR[14832] chan_sip.c: Unable to build sip pvt data for<br>
> 'TRUNK/DESTINATION' (Out of memory or socket error).<br>
><br>
> I had installed Asterisk-1.4.21, but this version stop from receiving calls<br>
> after these errors occured.<br>
><br>
> Then I downgrade to version 1.4.19 (because I had have tested that version),<br>
> but after getting these error it stop from creating the outbound call.<br>
><br>
> The configuration of the * is an incomming call from the my SIP Provider and<br>
> after internal manage it makes a second call to other destination--DID--.<br>
><br>
> For AGI compatibility issues I could not use Version 1.4.22 (issues whith<br>
> DeadAGI for billing purpuses).<br>
><br>
><br>
><br>
> This is my rtp.conf<br>
><br>
><br>
> [general]<br>
> ;<br>
> ; RTP start and RTP end configure start and end addresses<br>
> ;<br>
> ; Defaults are rtpstart=5000 and rtpend=31000<br>
> ;<br>
> rtpstart=10000<br>
> rtpend=20000<br>
><br>
><br>
> This is my sip.conf for the TRUNK<br>
><br>
><br>
> [TRUNK]<br>
> type=peer<br>
> nat=never<br>
> host=destination.public.ip.address<br>
> fromdomain=my.public.ip.address<br>
> dtmfmode=rfc2833<br>
> canreinvite=no<br>
> disallow=all<br>
> allow=g729<br>
><br>
><br>
> Please help.<br>
> --<br>
> Juan E. Rodríguez<br>
><br>
<br>
</div></div>Juan,<br>
<br>
You might need to increase the number of file descriptors available<br>
in Linux. What distro are you on? Are you using the Asterisk startup<br>
scripts? In later versions this is done for you automatically if you<br>
are running Asterisk as root. Have a look at this:<br>
<br>
<a href="http://www.voip-info.org/wiki/view/file+descriptors" target="_blank">http://www.voip-info.org/wiki/view/file+descriptors</a><br>
<br>
--<br>
Kristian Kielhofner<br>
<a href="http://blog.krisk.org" target="_blank">http://blog.krisk.org</a><br>
<a href="http://www.submityoursip.com" target="_blank">http://www.submityoursip.com</a><br>
<a href="http://www.astlinux.org" target="_blank">http://www.astlinux.org</a><br>
<a href="http://www.star2star.com" target="_blank">http://www.star2star.com</a><br>
<br>
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</blockquote></div><br><br clear="all"><br>-- <br>Juan E. Rodríguez<br>Cel. 829-886-5565<br>Work: 809-724-9227<br>
</div></div>