[asterisk-dev] UAC leg cancel on early media / MoH.

[asterisk at ntplx.net]

Most systems have an un-answered call timeout/limit. This makes sure
that calls are not left ringing forever. Also most telcos enforce a
timeout to prevent people from getting endless free phone calls.

If you want to do something with a call, other than leave it ringing,
then answer it first (as you found out). Some 800 call centers cheat the
phone company by NOT answering the call but starting an auto attendent or
voice mail system and placing the call in queue WITHOUT being charged.

There are valid reasons to not answer a call but not be ringing.
The best example is for changed/disconnected number messages.
Other reasons not to answer are for call forwarding/redirections
(where the final leg controls answering).

Also some phones (or systems) won't send DTMF on an unanswered call.....


Quoting Alex Balashov <abalashov at evaristesys.com>:

>
>
> Hi folks,
>
> I ran into a problem where SIP calls were being dumped straight into a queue
> without being Answer()'d.  Music on hold from the queue was being
> generated via 183 Session in Progress + SDP, i.e. early media / in-band
> ringback.
>
> After about 3 minutes of this, all SIP UACs I tested with would CANCEL
> the leg, resulting in the caller being dropped out of the queue.  This
> happened with a Cisco 7960 (SIP image), Polycom 501, and tne X-lite
> softphone.
>
> Anyway, I fixed the problem by simply furnishing an Answer() in the
> dial plan, of course, but I was curious as to why SIP UACs react this
> way.  I could not find any explanation for this in reviewing the
> various SIP T-timers in the RFC, or the various RFCs and drafts dealing with
> early media.
>
> In other words, I see no reason why the calling SIP agent should   
> terminate the
> call after 3 minutes since the 183 + SDP have elapsed.  What gives?
>
> Thanks,
>
> --
> Alex Balashov
> Evariste Systems
> Web    : http://www.evaristesys.com/
> Tel    : +1-678-954-0670
> Direct : +1-678-954-0671
>
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