Hello, I have question regarding the changes that are made in the sip protocol in Asterisk - the option progressinband. When this option is set to yes in asterisk version 1.4.21.1 - the call flow is:
sip.conf: progressinband=yes Device Asterisk -----------INVITE SDP---------> <---------100 Trying------------ <-----183 Session Prgoress-- After version 1.4.2X+ (tested with 1.4.36/1.4.41.1) the call flow changes to: Device Asterisk -----------INVITE SDP---------> <---------100 Trying------------ <---------180 Ringing---------- <-----183 Session Prgoress-- >From the information that I was able to found in the last version it will always send 180 Ringing and after that the 183 Session Progress. Is there any way to configure the system to behave as in version 1.4.21.1? The show stopper for me is the new behaver. I known that - there is no problem to send 180 Ringing before sending the 183 Session Progress and if there is a problem it is in the other device, not in the Asterisk. Also is the new behaver (1.4.2X+) presented in all new versions of Asterisk 1.6+/1.8+? Regards, Anatoliy -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
