[asterisk-dev] Bounty

[CDR]

I agree. The timer would wait for a SIP 180 or 100 or 183, maybe
configurable in sip.conf?
I would not work with a carrier that does not provide some packet in this
manner.
Federico

On Wed, Feb 24, 2010 at 8:24 PM, Alex Balashov <abalashov at evaristesys.com>wrote:

> Who told you that you're always going to get a 180 Ringing reply?
>
> Most providers that provide PSTN trunking will give you ringback as
> in-band via an early dialog (183 Session in Progress).  With some
> calls, you may just get a provisional 100 Trying reply and then
> nothing until a sudden 200 OK.  There are many possible flows and
> scenarios.
>
> On 02/24/2010 07:59 PM, CDR wrote:
>
> > I need a new Timeout parameter added to the Dial application, for SIP
> > dialing. The new timeout would be "first-ring" timeout, as opposed to
> > timeout for connection. If we don't get a 180 Ringing message before a
> > certain amount of seconds, the call fails. This a needed addition to
> > Asterisk. I need this in version 1.4 and cannot wait the normal time for
> > a "new feature" process to complete. The rationale is clear: many
> > carriers will hold the call indefinitely, instead of returning a 503. If
> > the call is ringing, then I don't care if it rings for 60 seconds, but
> > if there is no ringback before 6 seconds, I need yo try another carrier
> > and move on.
> >
> > Please contact me at nine five four 444 seven 4 zero 8
> >
>
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
>
> Tel    : +1 678-954-0670
> Direct : +1 678-954-0671
> Web    : http://www.evaristesys.com/
>
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