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Dear All,
<p>I was wondering if this could be the right place to send a feature request
or not, but any way this something that I have dicussed with many of asterisk
implementers here in Egypt, and it is something we think would be a nice
addition to Asterisk.
<p>We have problem arising that there are so many codecs currently in use
and many want to use the same codecs and refuse to change it (especially
that g723 needs licence and gsm and other gx codecs are used instead),
and so we can find a number of asterisk boxes need to recieve and send
calls in different formats between each other, and at the same time need
to do a limitation of concurrent number of calls, in the past this was
easy you decide which code to use and you can calculate the number of calls
that equals the available bandwidth, but if you use different types of
codecs this will be hard and seems like shooting a star down, but if we
can say to asterisk what bandwidth is available for the box to use for
incoming and outside calls this will make things much smother and easier
to manage, this bandwidth feature is available in Cisco Call Manager, actually
it is an esstintial element in creating a sip trunk.
<br>
<p>I am ready to answer any questions or comments regarding this issue.
<br>
<pre>--
Thx
MAG</pre>
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