Please start new threads for new messages (don't "reply" and just wipe out the body). The headers still exist so you wind up with screwy threading in the list archives (ditto for those of us who have e-mail software that supports threading).
AR On 7/31/07, Richard Brady <[EMAIL PROTECTED]> wrote: > > 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? > > 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 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Alex Robar [EMAIL PROTECTED]
_______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
