<div dir="ltr"><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px; ">After getting some ERRORS like this:<br><br>[Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call.<br>
[Oct 8 21:42:49] ERROR[2485] rtp.c: No RTP ports remaining. Can't setup media stream for this call.<br>[Oct 8 21:42:49] ERROR[31903] rtp.c: No RTP ports remaining. Can't setup media stream for this call.<br>[Oct 8 21:42:49] ERROR[2489] rtp.c: No RTP ports remaining. Can't setup media stream for this call.<br>
<br>I start getting:<br><br>ERROR[14844] chan_sip.c: Unable to build sip pvt data for '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 'TRUNK/DESTINATION' (Out of memory or socket error).<br>
<br>I had installed Asterisk-1.4.21, but this version stop from receiving calls after these errors occured.<br><br>Then I downgrade to version 1.4.19 (because I had have tested that version), but after getting these error it stop from creating the outbound call.</span><div>
<span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><br></span></div><div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;">The configuration of the * is an incomming call from the my SIP Provider and 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 DeadAGI for billing purpuses).<br></span></div><div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><br>
</span></div><div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><br></span></div><div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;">This is my rtp.conf</span></div>
<div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><br></span></div><div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><div>
[general]</div><div>;</div><div>; RTP start and RTP end configure start and end addresses</div><div>;</div><div>; Defaults are rtpstart=5000 and rtpend=31000</div><div>;</div><div>rtpstart=10000</div><div>rtpend=20000</div>
<div><br></div></span><div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><br></span></div><div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;">This is my sip.conf for the TRUNK</span></div>
<div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><br></span></div><div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><div>
[TRUNK]</div><div>type=peer</div><div>nat=never</div><div>host=destination.public.ip.address</div><div>fromdomain=my.public.ip.address</div><div>dtmfmode=rfc2833</div><div>canreinvite=no</div><div>disallow=all</div><div>allow=g729</div>
<div><br></div></span></div><div><span class="Apple-style-span" style="font-family: Verdana; -webkit-border-horizontal-spacing: 1px; -webkit-border-vertical-spacing: 1px;"><br clear="all"></span>Please help.<br>-- <br>Juan E. Rodríguez<br>
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