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

Reply via email to