<div dir="ltr">Hi Kevin, <div><br></div><div><a href="http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension">http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension</a> <br clear="all">
<div dir="ltr"><br></div><div dir="ltr">this says 4 active ports for one call</div><div dir="ltr"><br></div><div dir="ltr">Regards,<br>Zohair Raza<div><br></div></div>
<br><br><div class="gmail_quote">On Wed, Feb 22, 2012 at 4:38 PM, Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="im">On 02/22/2012 06:26 AM, virendra bhati wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Does anyone know the correct information of my question. All are move<br>
round and round .<br>
</blockquote>
<br></div>
What does that mean? I answered your question with the correct and complete information.<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="im">
<br>
On Tue, Feb 21, 2012 at 7:28 PM, Kevin P. Fleming <<a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a><br></div><div class="im">
<mailto:<a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a>>> wrote:<br>
<br>
On 02/21/2012 07:51 AM, Alex Balashov wrote:<br>
<br>
As many ports as required by the nature of the call, i.e. the<br>
protocol(s) used for the bearer.<br>
<br>
<br>
For an IAX2 call, the answer is 'zero' for all of those call types<br>
(at least the ones that are supported in IAX2, not all of them are).<br>
<br>
For protocols that use RTP for media transport, two ports are<br>
required for each media stream (one for RTP, one for RTCP).<br>
</div></blockquote><div class="HOEnZb"><div class="h5">
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
Jabber: <a href="mailto:kfleming@digium.com" target="_blank">kfleming@digium.com</a> | SIP: <a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a> | Skype: kpfleming<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
--<br>
______________________________<u></u>______________________________<u></u>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/<u></u>mailman/listinfo/asterisk-<u></u>users</a><br>
</div></div></blockquote></div><br></div></div>