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<DIV class=OutlookMessageHeader align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006>Phones are not behind
NAT. </SPAN></FONT></DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006></SPAN></FONT> </DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006>Every client is on the same internal network as
the asterisk pbx (nothing is sent through the internet). It's not the
network since I tested this by calling asterisk from an outside phone (cell) and
let asterisk play a message for me. Same "cutting" and "chopping" when many
SIP-clients where active in a call at the same time.</SPAN></FONT></DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006></SPAN></FONT> </DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006>Computer RAM is 2 gb.</SPAN></FONT></DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006></SPAN></FONT> </DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006>If the E1 is channelized or not I don't actually know.
How would I know this and why would it affect the call quality when many people
are in a call at the same time (same lines work fine with an Ericsson
BusinessPhone Exchange)?</SPAN></FONT></DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006></SPAN></FONT> </DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006>Thanks!</SPAN></FONT></DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006></SPAN></FONT> </DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006>Regards,</SPAN></FONT></DIV>
<DIV align=left><FONT face="Courier New" size=2><SPAN class=921523314-04072006>Jan</SPAN></FONT></DIV><BR>
<DIV class=OutlookMessageHeader align=left>
<HR>
<FONT face=Tahoma size=2><B>Från:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>För
</B>broadbandvoice@comcast.net<BR><B>Skickat:</B> den 4 juli 2006
15:55<BR><B>Till:</B> Asterisk Users Mailing List - Non-Commercial
Discussion<BR><B>Ämne:</B> Re: SV: [Asterisk-Users] Running 40 active calls (too
much för CPU?)<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV>Are the phones behind a NAT? What is the processory memory size? Are the E1
channelized?</DIV>
<DIV> </DIV>
<BLOCKQUOTE style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #1010ff 2px solid">--------------
Original message -------------- <BR>From: <jan.sarin@securia.se>
<BR><BR>> I should add that thease 25 calls where SIP (internal) to Zap
(PSTN) calls. <BR>> <BR>> Mvh, <BR>> Jan <BR>> <BR>>
-----Ursprungligt meddelande----- <BR>> Från:
asterisk-users-bounces@lists.digium.com <BR>>
[mailto:asterisk-users-bounces@lists.digium.com] För jan.sarin@securia.se
<BR>> Skickat: den 4 juli 2006 09:41 <BR>> Till:
asterisk-users@lists.digium.com <BR>> Ämne: [Asterisk-Users] Running 40
active calls (too much för CPU?) <BR>> <BR>> Hi, <BR>> <BR>> We're
running asterisk 1.2.1 on a Dell PowerEdge 600SC (2.4 ghz) server <BR>>
connected to the PSTN through two E1 pipes to a TE405P. This has been running
<BR>> just fine for several months... <BR>> <BR>> But yesturday we
connected a large number of softphone SIP clients (50) and 25 < BR>>
; of these where running simultaneous active calls on the INTERNAL ethernet
using <BR>> g711 (ulaw). We noticed that the sound was jagged just as if
the CPU couldn't <BR>> handle 25 calls (?!). <BR>> <BR>> I checked
the CPU load and it never went over 55 % and memusage was low too. <BR>>
<BR>> Does anyone know what could be the problem? Are there some kind of
CPU spikes <BR>> that make these cuts in the audio? If so, why on earth
can't a 2,4 ghz processor <BR>> handle 25 low-quality audio "tracks" on
asterisk when I can run +50 cd-quality <BR>> audio tracks when producing
music? <BR>> <BR>> ANY help and/or comments would be appreciated since
this is quite an acute <BR>> problem. <BR>> <BR>> Regards, <BR>>
Jan <BR>> _______________________________________________ <BR>>
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