[asterisk-dev] Feature Request: Assigning number of calls accepted by Bandwidth

[Mohamed A. Gombolaty]

Dear All,

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.

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.


I am ready to answer any questions or comments regarding this issue.


--
Thx
MAG


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