[asterisk-users] When should a Progress or Ringing be used in a today's telephony ?

Daniel Tryba daniel at tryba.nl
Wed May 16 10:59:04 CDT 2018

On Wed, May 16, 2018 at 04:51:49PM +0200, Olivier wrote:
> 1. When Asterisk receives a SIP call coming from PSTN, is there a time
> frame within which Asterisk must reply something to keep caller from
> canceling the call ? Where does this limit come from ? From SIP RFC ? From
> local regulation bodies ?
> 2. Which SIP signal is required to stop call cancellation in the previous
> case ?

See RFC 3261, 17.1.1. A (provisional) response to an INVITE is required
within a timelimit. After a provisional response a non-provisional
response is required. Defaults are on page 264 of the RFC (first to
> 3. When Asterisk receives a call, either from PSTN or from a SIP phone) it
> cannot process (unkown callee, whatever reason, ...), should you stop
> processing with Hangup or Congestion ?
> Hangup application allow for exit code customization, if I'm not mistaken,
> but  Congestion exists for a reason.

With regard to PSTN calls the signalig is limited, but to a SIP device
you could signal usefull information, eg: unknown, temp. unavailable.
Why not give a usefull reason instead of Congestion

> 4. Is it a good practise to send a 180/183 when you don't get one ?

People will complain if there is no indication, so yes IMHO.

> 5. I observed I sometimes got  a 100 Trying then a 183 session Progress
> when outcalling some (mobile) phones while simpy getting 100 Trying when
> some other (mobile) phone through the same carrier (most probably, end
> devices were not managed by the same (mobile) telephony provider).
> What explains such difference ?

An explanation could be packet loss. But there are no requirements for
1xx responses to an INVITE. Maybe they just don't care about feedback to

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