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 last). > 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 callers. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
