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On 11/02/2011 04:13 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);">150/4 = 37.5. maybe your sip peer has a
conflicting range?</span></p>
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</blockquote>
<br>
Where do I set this range in my peer definition ? I don't think
there is such a parameter in sip.conf<br>
<br>
<br>
To be perfectly clear, how many RTP-ports are needed in the below
situation :<br>
<br>
- an incoming call to a group of SIP-peers (10 in total)<br>
- 1 peer answers this incoming call<br>
<br>
My thought : 2 RTP for incoming channel, 2 RTP for channel to SIP
peer<br>
(and the other peers don't matter)<br>
<br>
Am I correct ?<br>
<br>
Or is there a need for a channel to every peer that is "ringing" ?<br>
<br>
<br>
<br>
<br>
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<p class="MsoNormal"><span style="font-size: 11pt; font-family:
"Calibri","sans-serif"; color: rgb(31,
73, 125);"><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 10:06 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial
Discussion<br>
<b>Subject:</b> Re: [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,<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>
</span><br>
On 11/02/2011 04:00 PM, Danny Nicholas wrote: <o:p></o:p></p>
<p class="MsoNormal"><span style="font-size: 11pt; font-family:
"Calibri","sans-serif";">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.</span><o:p></o:p></p>
<p class="MsoNormal"><span style="font-size: 11pt; font-family:
"Calibri","sans-serif";"> </span><o:p></o:p></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 moz-do-not-send="true"
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a moz-do-not-send="true"
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</span><o:p></o:p></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>
<pre><o:p> </o:p></pre>
<pre><o:p> </o:p></pre>
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