In fact as far as I know, Asterisk stands in the middle of calls,
breaking one transaction and initiating another to the other side, doing
the bridge between them... Although good in some cases like permitting
to start a new transaction to the next hop changing codecs, in my case I
don't need that feature because I'm using reINVITEs to implement
session-timer support in the user agent to solve problems of whong
accounting if power failure or link happens...
Is there any way to disable those breaks in audio stream?
Regards,
Ricardo.
Andreas Sikkema wrote:
My Asterisk server is working fine, although every time that
in the middle of
any call there is a reinvite, the user hears a glitch. Why is
this happening?
How can I solve this problem?
That's because a REINVITE is generally used to change from one
codec to another. For some reason this involves stopping the
existing audio, waiting a little while and then starting a new
audio stream.
So far this one of the reasons why I don't like reinvite...
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