http://www.ietf.org/rfc/rfc3665.txt Session Initiation Protocol (SIP) Basic Call Flow Examples
Jim James H. Thompson [EMAIL PROTECTED] ----- Original Message ----- From: "Rich Adamson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, February 04, 2004 9:41 AM Subject: RE: [Asterisk-Users] Sip flow diagram? Answers > For the purposes of future archive searches, the following is a > summary from those that responded. Thanks very much to everyone!!!! > > http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programming_reference_guide_book0 > 9186a0080080221.html > http://www.faqs.org/rfcs/rfc3261.html > http://www.iptel.org/info/players/ietf/callflows/ > > ------------------------ > > > > > > Does anyone have a high level flow diagram showing acceptable sip > > > messages exchanges? > > > > > > For exampe: > > > Source Dest > > > Invite -> > > > <- Trying > > > Ok -> > > > > > > I'm specifically trying to debug an issue with various hangups, prior > > > to call completion, after call completion, "calling" vs "called" party > > > hold, etc, and getting rather confused watching the various packets > > > flowing between sip devices with a sniffer (and no reference document). > > > > > > Rich > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
