[asterisk-dev] SIP Channel does not release.

[Anton]

Hello,
Just to point out to the SIP issues which exist in 1.4svn (up to SVN-branch-1.4-r46663) (and possible trunk). 
When SIP call is made from one ASTERISK box to another - 
and second box RINGING, if we hangup the caller station, 
without answering callee, callee still rings. channel list gives that call is active on the callee asterisk, 
while calling station has no calls in cahnnel list.
Seems SIP does not send release.

SIP/mediagw1-0825cfb 935055490 at transit:1  Down    AppDial((Outgoing Line))
SIP/192.168.1.35-082 935055490 at transit:1  Ring    Dial(SIP/935055490 at mediagw1)

with IAX2 this does not happen.

Regards,