]> git.ipfire.org Git - thirdparty/asterisk.git/commit
Fix an issue where cancelled outgoing SIP calls would erroneously report the device...
authorMark Michelson <mmichelson@digium.com>
Thu, 19 Mar 2009 16:04:02 +0000 (16:04 +0000)
committerMark Michelson <mmichelson@digium.com>
Thu, 19 Mar 2009 16:04:02 +0000 (16:04 +0000)
commit338e48e055627d1f999bef275d0fe02d12e6fd68
treea493dd6498771fb8df2c89f7c6cf981f568165c5
parent3849208a9980b1801ac8ded8ed4b8b4f2c87c0e9
Fix an issue where cancelled outgoing SIP calls would erroneously report the device as "in use."

A user was having an issue where if an outgoing SIP call was canceled, the SIP device
would remain in use if we had not received any response to the initial INVITE we sent out.
The SIP device would remain in use until the autocongestion timer was exhausted.

I tracked down the cause of this to be the section of code I am removing here. I asked several
people what the purpose of this code was meant to be, but no one could give me any sort of
answer as to why this was here. The person who was having this issue has been using this patch
for several months and it has stopped the problems they have had.

AST-196

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183115 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c