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(${ARG1},300,R)
-> INVITE
<- 100 Trying
<- 180 Ringing
<- 183 Session Progress
( <- RTP -> )
<- 200 OK
any ideas why Ringing is missing? any solutions?
Marek
On Tue, Dec 1, 2020 at 7:20 AM marek <cervajs64 at gmail.com> wrote:> hi, > > after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know, > its old. customer is very conservative...) > > i have problem with missing 180 Ringing > > flow is easy (PBX -> Asterisk -> SIP SBC) > > Asterisk 11 > PBX - Asterisk > -> INVITE > <- 100 Trying > <- 183 Session Progress > ( <- RTP -> ) > <- 180 Ringing > <- 200 OK > > Asterisk 13 > -> INVITE > <- 100 Trying > <- 183 Session Progress > ( <- RTP -> ) > > __MISSING RINGING___ > > <- 200 OK > > temporarily i solved problem with using "R" param > > R: Default: Indicate ringing to the calling party, even if the called party > isn't actually ringing. Allow interruption of the ringback if early > media > is received on the channel. > > it changed to > > Asterisk 13 (Dial(${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>
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-bounces at lists.digium.com im Auftrag von cervajs64 at
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