<div dir="ltr">Each asterisk has its own service...Asterisk1 has asterisk1 as service...If you want to enter to asterisk1 console, you just type asterisk1 -rvvvv and so on...<br><br>regarding IPs, I have created different virtual Network interfaces on the same real Interface and assign an Ip to each instance of asterisk<br>
<br>Regards<br><br><div style="visibility: hidden; display: inline;" id="avg_ls_inline_popup"></div><style type="text/css">#avg_ls_inline_popup { position:absolute; z-index:9999; padding: 0px 0px; margin-left: 0px; margin-top: 0px; width: 240px; overflow: hidden; word-wrap: break-word; color: black; font-size: 10px; text-align: left; line-height: 13px;}</style><div class="gmail_quote">
On Mon, Jun 21, 2010 at 12:34 AM, CDR <span dir="ltr"><<a href="mailto:venefax@gmail.com">venefax@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Open VZ is much lighter than ESX.<br>The question is how do you enter only one instance of asterisk, like "asterisk -r". I assume that each instance of Asterisk is using a different IP address in sip.conf<br>Yours<br>
F.A.<div><div></div><div class="h5"><br><br><div class="gmail_quote">On Sun, Jun 20, 2010 at 3:31 PM, michel freiha <span dir="ltr"><<a href="mailto:michofr@gmail.com" target="_blank">michofr@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div dir="ltr">Hello Philip,<br><br>Regarding installation, I have installed each asterisk server on a different folder...I Installed asterisk1 under /usr/local/asterisk1 and asterisk2 on /usr/local/asterisk2 and so on...Even libraries are installed on different folders...Everything is OK but I'm facing this voice quality issue...Can I ask you why you used Virtualization solution? What was your issue?<br>
<br>Currently, I'm using VMWare (ESXI)...It's a great solution but sometimes I have an issue in one Virtual machine...It stuck and goes down...That's why I decided to use multiple asterisk on same OS<br><br>Regards<div>
<div></div><div><br>
<br><br><div class="gmail_quote">On Sat, Jun 19, 2010 at 12:32 PM, CDR <span dir="ltr"><<a href="mailto:venefax@gmail.com" target="_blank">venefax@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Could you send me details of your installation?<br>I have tried the same set up but ended up by using Virtuozzo (OpenVZ). <br>Philip<br><br><div class="gmail_quote"><div><div></div><div>On Sat, Jun 19, 2010 at 5:21 AM, michel freiha <span dir="ltr"><<a href="mailto:michofr@gmail.com" target="_blank">michofr@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div><div dir="ltr">Dear All,<br>
<br>
I have installed 4 asterisks on the same Centos machine..>Each
Asterisk has its own installation folder and use its own
libraries...Everything looks great and all asterisks are doing their
jobs correctly except one thing...I faced a voice quality issue...On a
specific time, and after the number of calls begin increasing, the
voice quality will begin degradation...<br>
<br>
Could it be a buffer issue that cause this voice quality problem or
something else?<br>
<br>
Waiting your reply<br>
<br>
Regards <div style="display: inline;"></div></div>
<br></div></div>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br></blockquote></div><br>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br></blockquote></div><br><div style="display: inline;">
</div></div></div></div>
<br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br></blockquote></div><br>
</div></div><br>--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br></blockquote></div><br></div>