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You need to decide which codecs you are going to allow to be used on
the SIP side. As you are connecting to E1 then the standard codec
would be g711 alaw or ulaw. You could force the SIP side to use the
same codec but it uses about 100Kbps of bandwidth so quite a bit
higher than other codecs and some customers may prefer to use a
higher compression codec so they dont have to have a faster internet
connection.<br>
If you start allowing other codecs then asterisk will need to
perform codec translation which consumes CPU.<br>
<br>
You will be limited by the E1 cards you can fit into asterisk.
Generally its best to keep to a maximum of 2 PCI cards in a server
as they each generate a lot of interrupts. A couple of Sangoma A116
will give you 32 E1 circuits which can handle 960 calls. That may
well be pushing the server to its limits of what it can do even if
it is just handling basic calls.<br>
So you will want 3 of these systems and preferably four as you will
want some redundancy.<br>
<br>
You would want something like opensips at the front end on a couple
of redundant servers. You can use the load balancing module in
opensips to spread the calls evenly between the available asterisk
servers.<br>
<br>
You should also have a think about whether you wish to use asterisk
at all. Opensips will do the load balancing that you need and can
perform the billing actions aswell. If all asterisk is doing is
performing sip to E1 conversion there may be hardware solutions
which will be more cost effective and require less maintenance.<br>
<br>
<br>
On 20/12/13 10:08, bilal ghayyad wrote:
<blockquote
cite="mid:1387534110.32473.YahooMailNeo@web160802.mail.bf1.yahoo.com"
type="cite">
<div style="color:#000; background-color:#fff;
font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial,
Lucida Grande, sans-serif;font-size:12pt">
<div>Thanks a lot for the help from the all.</div>
<div><br>
</div>
<div style="color: rgb(0, 0, 0); font-size: 16px; font-family:
HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; background-color: transparent;
font-style: normal;">Without using PBX functionalities (like
conference or pickup and so on), only basic calls. So how many
concurrent calls can support?</div>
<div style="color: rgb(0, 0, 0); font-size: 16px; font-family:
HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; background-color: transparent;
font-style: normal;"><br>
</div>
<div style="color: rgb(0, 0, 0); font-size: 16px; font-family:
HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; background-color: transparent;
font-style: normal;">The idea is, we need to use Asterisk with
ISP which will be service provider for sip calls for the
subscribers, and the asterisk should be connected with E1s to
do calls within the country.</div>
<div style="color: rgb(0, 0, 0); font-size: 16px; font-family:
HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; background-color: transparent;
font-style: normal;"><br>
</div>
<div style="color: rgb(0, 0, 0); font-size: 16px; font-family:
HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; background-color: transparent;
font-style: normal;">The registered users will reach up to 100
000 users and the concurrent calls will reach up to 2000 or
3000 calls.</div>
<div style="color: rgb(0, 0, 0); font-size: 16px; font-family:
HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; background-color: transparent;
font-style: normal;"><br>
</div>
<div style="color: rgb(0, 0, 0); font-size: 16px; font-family:
HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; background-color: transparent;
font-style: normal;">Appreciate the kindly advise.</div>
<div style="color: rgb(0, 0, 0); font-size: 16px; font-family:
HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; background-color: transparent;
font-style: normal;">Regards</div>
<div style="color: rgb(0, 0, 0); font-size: 16px; font-family:
HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; background-color: transparent;
font-style: normal;">Bilal</div>
<div class="yahoo_quoted" style="display: block;"> <br>
<br>
<div style="font-family: HelveticaNeue, 'Helvetica Neue',
Helvetica, Arial, 'Lucida Grande', sans-serif; font-size:
12pt;">
<div style="font-family: HelveticaNeue, 'Helvetica Neue',
Helvetica, Arial, 'Lucida Grande', sans-serif; font-size:
12pt;">
<div dir="ltr"> <font face="Arial" size="2"> On
Wednesday, December 18, 2013 6:10 PM, Tech Support
<a class="moz-txt-link-rfc2396E" href="mailto:asterisk@voipbusiness.us"><asterisk@voipbusiness.us></a> wrote:<br>
</font> </div>
<div class="y_msg_container">
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<div class="yiv9305735573WordSection1">
<div class="yiv9305735573MsoNormal"><span
style="font-size:11.0pt;">You can have tens of
thousands of phones as long as no one makes or
receives any calls </span><span
style="font-size: 11pt; font-family:
Wingdings; color: rgb(31, 73, 125);">J</span><span
style="font-size:11.0pt;">. The better
question to ask is how many concurrent calls
have people been able to make. The quick
answer is it depends on many things. </span></div>
<div class="yiv9305735573MsoNormal"><span
style="font-size:11.0pt;">John </span></div>
<div class="yiv9305735573MsoNormal"><span
style="font-size:11.0pt;"> </span></div>
<div class="yiv9305735573yqt1908056049"
id="yiv9305735573yqt07171">
<div>
<div style="border:none;border-top:solid
#B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in;">
<div class="yiv9305735573MsoNormal"><b><span
style="font-size:10.0pt;">From:</span></b><span
style="font-size:10.0pt;">
<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>bilal ghayyad<br
clear="none">
<b>Sent:</b> Wednesday, December 18,
2013 9:46 AM<br clear="none">
<b>To:</b> Asterisk Users Mailing List -
Non- Commercial Discussion<br
clear="none">
<b>Subject:</b> [asterisk-users] Maximum
number of users</span></div>
</div>
</div>
<div class="yiv9305735573MsoNormal"> </div>
<div>
<div>
<div class="yiv9305735573MsoNormal"
style="background:white;"><span style="">Hello;</span></div>
</div>
<div>
<div class="yiv9305735573MsoNormal"
style="background:white;"><span style="">
</span></div>
</div>
<div>
<div class="yiv9305735573MsoNormal"><span
style="">Can someone advise me what is
the maximum number of users (IP Phones)
that can be supported by asterisk 1.8 or
later?</span></div>
</div>
<div>
<div class="yiv9305735573MsoNormal"><span
style=""> </span></div>
</div>
<div>
<div class="yiv9305735573MsoNormal"><span
style="">Regards</span></div>
</div>
<div>
<div class="yiv9305735573MsoNormal"><span
style="">Bilal</span></div>
</div>
</div>
</div>
</div>
</div>
</div>
<br>
<br>
</div>
</div>
</div>
</div>
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