On Tue, Dec 1, 2020 at 7:20 AM marek <cervajs64 at 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(${ARG1},300,R) > -> INVITE > <- 100 Trying > <- 180 Ringing > <- 183 Session Progress > ( <- RTP -> ) > <- 200 OK > > any ideas why Ringing is missing? any solutions? >Have you compared the signaling in both directions between the two versions to see if there is a difference? -- Joshua C. Colp Asterisk Technical Lead Sangoma Technologies Check us out at www.sangoma.com and www.asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20201201/29d9c1be/attachment.html>
Dne 01/12/2020 v 12:58 Joshua C. Colp napsal(a):> On Tue, Dec 1, 2020 at 7:20 AM marek <cervajs64 at gmail.com > <mailto:cervajs64 at gmail.com>> wrote: > > hi, > > after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know, > its old. customer is very conservative...) > > i have problem with missing 180 Ringing > > flow is easy (PBX -> Asterisk -> SIP SBC) > > Asterisk 11 > PBX - Asterisk > -> INVITE > <- 100 Trying > <- 183 Session Progress > ( <- RTP -> ) > <- 180 Ringing > <- 200 OK > > Asterisk 13 > -> INVITE > <- 100 Trying > <- 183 Session Progress > ( <- RTP -> ) > > __MISSING RINGING___ > > <- 200 OK > > temporarily i solved problem with using "R" param > > R: Default: Indicate ringing to the calling party, even if the > called party > isn't actually ringing. Allow interruption of the ringback if > early > media > is received on the channel. > > it changed to > > Asterisk 13 (Dial(${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) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20201201/5242e789/attachment.html>