[Asterisk-Dev] SIP bug

[Duane]

Friend and I updated CVS versions of our asterisk servers and well 
something bad happened with chan_sip...

Current CVS to anything else seems to be ok... Current CVS  to Current 
CVS after about a minute sounds like the loop current dropped out even 
though we're both using SIP hardphones...

If we connect to each other over IAX2 this problem isn't exibited...

-- 
Best regards,
  Duane

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