On Tue, Dec 1, 2020 at 11:24 AM marek <cervaj...@gmail.com> wrote: > > Dne 01/12/2020 v 12:58 Joshua C. Colp napsal(a): > > On Tue, Dec 1, 2020 at 7:20 AM marek <cervaj...@gmail.com> wrote: > >> 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? >> > > Have you compared the signaling in both directions between the two > versions to see if there is a difference? > > whats your goal with this question? > > asking if there are some side effects in incoming call ? (SBC -> Asterisk > -> PBX) >
No side effects, but looking at the actual SIP signaling (sip set debug on) and see what the remote side is sending for SIP responses as well. -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org
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