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"I guess that having only ip-phones in mind is not a good approach. Since it is possible to have a sip proxy connected to asterisk we could receive a 407 (unauthorized) or 483 (too many hops) as response and dialog ending would not be a good behavior." So modified. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160480 65c4cc65-6c06-0410-ace0-fbb531ad65f3