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<font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
thank you for your answer...<br>
<br>
Current range (rtp.conf) : 11500 - 11650<br>
<br>
Current calls : 20 à 25<br>
<br>
Is this not sufficient ??<br>
<br>
<br>
<br>
<br>
Jonas.<br>
<br>
<br>
</font><br>
On 11/02/2011 04:00 PM, Danny Nicholas wrote:
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<p class="MsoNormal"><span style="font-size: 11pt; font-family:
"Calibri","sans-serif"; color: rgb(31,
73, 125);">You have set an insufficient range in rtp.conf.
Asterisk uses 2 ports per call, but allocates 4 for
transferring, etc, so when you set up a range of 10001-10040
(for example) you are basically setting a range of 10
concurrent calls. Check rtp.conf and make the end range
larger by 8 or 12 or whatever number of extra calls you’d
like to see before you get this message again.<o:p></o:p></span></p>
<p class="MsoNormal"><span style="font-size: 11pt; font-family:
"Calibri","sans-serif"; color: rgb(31,
73, 125);"><o:p> </o:p></span></p>
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<p class="MsoNormal"><b><span style="font-size: 10pt;
font-family:
"Tahoma","sans-serif"; color:
windowtext;">From:</span></b><span style="font-size:
10pt; font-family:
"Tahoma","sans-serif"; color:
windowtext;"> <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] <b>On
Behalf Of </b>Jonas Kellens<br>
<b>Sent:</b> Wednesday, November 02, 2011 9:57 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial
Discussion<br>
<b>Subject:</b> [asterisk-users] Unable to build sip pvt
data - Switching equipment congestion<o:p></o:p></span></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="font-family:
"Helvetica","sans-serif";">Hello list,<br>
<br>
can anyone tell me what the following means (found in
messages log) :<br>
<br>
<br>
<i>[Nov 2 11:16:21] ERROR[18407] rtp.c: No RTP ports
remaining. Can't setup media stream for this call.<br>
[Nov 2 11:16:21] WARNING[18407] chan_sip.c: Unable to
create RTP audio session: Address already in use<br>
[Nov 2 11:16:21] ERROR[18407] chan_sip.c: Unable to build
sip pvt data for 'sipaccount7' (Out of memory or socket
error)<br>
[Nov 2 11:16:21] WARNING[18407] app_dial.c: Unable to
create channel of type 'SIP' (cause 42 - Switching
equipment congestion)</i><br>
<br>
<br>
Thank your for explaining the problems and a possible
solution !<br>
<br>
<br>
Greetingz,<br>
Jonas.</span><o:p></o:p></p>
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