If you have 183 Session progress, there is no need to send 180 Ringing (especially not AFTER 183 Session progress), as you already have early media instead. Having both is actually a bit misleading IMHO.
So this is actually correct. One should not rely on any of these 1xx "Provisional" messages. They may or may not be sent, without violating SIP standards. Am 01.12.20, 12:20 schrieb "asterisk-users im Auftrag von marek" <asterisk-users-boun...@lists.digium.com im Auftrag von cervaj...@gmail.com>: hi, after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know, its old. customer is very conservative...) i have problem with missing 180 Ringing flow is easy (PBX -> Asterisk -> SIP SBC) Asterisk 11 PBX - Asterisk -> INVITE <- 100 Trying <- 183 Session Progress ( <- RTP -> ) <- 180 Ringing <- 200 OK Asterisk 13 -> INVITE <- 100 Trying <- 183 Session Progress ( <- RTP -> ) __MISSING RINGING___ <- 200 OK temporarily i solved problem with using "R" param R: Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Allow interruption of the ringback if early media is received on the channel. it changed to Asterisk 13 (Dial(${ARG1},300,R) -> INVITE <- 100 Trying <- 180 Ringing <- 183 Session Progress ( <- RTP -> ) <- 200 OK any ideas why Ringing is missing? any solutions? Marek -- _____________________________________________________________________ -- 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