Richard Brady
2007-Jul-31 13:26 UTC
[asterisk-users] Turn off SIP 183 Session Progress in Asterisk 1.4.8
[Resent due to non-descriptive subject line.] Hi folks When connecting two SIP users, is there any way to stop Asterisk from sending SIP 183 Session Progress messages, either globally or per-peer? Scenario as follows: Call from UA1 to Asterisk (UA2) to UA3. UA3 sends RTP before SIP OK to Asterisk (UA2). Asterisk (UA2) detects early audio from UA3 and sends 183 Session Progress with SDP to UA1. Instead I would like it to just send on the early audio, is this possible? Thanks in advance, Richard
Andrew Joakimsen
2007-Jul-31 23:36 UTC
[asterisk-users] Turn off SIP 183 Session Progress in Asterisk 1.4.8
On 7/31/07, Richard Brady <rnbrady at gmail.com> wrote:> > [Resent due to non-descriptive subject line.] > > Hi folks > > When connecting two SIP users, is there any way to stop Asterisk from > sending SIP 183 Session Progress messages, either globally or > per-peer?Yes, the option is progressinband in sip.conf, see: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband Scenario as follows:> Call from UA1 to Asterisk (UA2) to UA3. > UA3 sends RTP before SIP OK to Asterisk (UA2). > Asterisk (UA2) detects early audio from UA3 and sends 183 Session > Progress with SDP to UA1. > > Instead I would like it to just send on the early audio, is this possible?Why do you care about the 183 or not? Because ignoring anything you said about SIP 183 what you want is to send SIP 183 which would indicate there is inband indications. See: http://www3.ietf.org/proceedings/99jul/slides/mmusic-sip183-99jul/index.htm Also it wouldnt hurt to read the SIP RFC's to have a better understanding of what is going on: ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt ftp://ftp.rfc-editor.org/in-notes/rfc2543.txt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070731/d7ab4f16/attachment.htm