Matthew Jordan 4b3476d016 Do not send a BYE when a provisional response arrives during a re-INVITE ...

Commits r369557 and r369579 were done to improve handling of re-INVITEs
when the UA that was supposed to receive the re-INVITE fails to respond.
A limitation of those patches occurred when a UA sent a provisional
response to the re-INVITE.  This triggered a sending of a BYE in
check_pending.  This patch tweaks the handling of the re-INVITE such that
a BYE is not sent in response to those messages.

(issue ASTERISK-19992)
Reported by: Steve Davies
Tested by: Steve Davies
patches:
  (reinvite_tweak.diff license #5012 by Steve Davies)
........

Merged revisions 369626 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 369627 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369628 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2012-07-05 17:03:43 +00:00

..

h323

Asterisk media architecture conversion - no more format bitfields

2011-02-03 16:22:10 +00:00

misdn

Multiple revisions 369001-369002

2012-06-15 16:20:16 +00:00

sip

Better handle re-INVITEs with provisional but no final repsonses

2012-07-03 14:49:19 +00:00

chan_agent.c

Unique Call ID logging Phases III and IV

2012-06-26 21:45:22 +00:00

chan_alsa.c

Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

2012-05-14 19:44:27 +00:00

chan_bridge.c

Finalize ast_channel opaquification

2012-03-13 18:20:34 +00:00

chan_console.c

Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

2012-05-14 19:44:27 +00:00

chan_dahdi.c

Unique Call ID logging Phases III and IV

2012-06-26 21:45:22 +00:00

chan_gtalk.c

Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

2012-05-14 19:44:27 +00:00

chan_h323.c

Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

2012-05-14 19:44:27 +00:00

chan_iax2.c

Unique Call ID logging Phases III and IV

2012-06-26 21:45:22 +00:00

chan_jingle.c

Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

2012-05-14 19:44:27 +00:00

chan_local.c

Hangup handlers - Dialplan subroutines that run when the channel hangs up.

2012-06-29 17:02:32 +00:00

chan_mgcp.c

Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

2012-05-14 19:44:27 +00:00

chan_misdn.c

Fix chan_misdn compile error.

2012-06-20 21:32:40 +00:00

chan_multicast_rtp.c

Opaquification for ast_format structs in struct ast_channel

2012-02-24 00:32:20 +00:00

chan_nbs.c

Opaquification for ast_format structs in struct ast_channel

2012-02-24 00:32:20 +00:00

chan_oss.c

Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

2012-05-14 19:44:27 +00:00

chan_phone.c

Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

2012-05-14 19:44:27 +00:00

chan_phone.h

rename DialTone.h to chan_phone.h because chan_phone.c is the only file using it

2008-05-28 20:00:21 +00:00

chan_sip.c

Do not send a BYE when a provisional response arrives during a re-INVITE

2012-07-05 17:03:43 +00:00

chan_sip.exports.in

Merge changes dealing with support for Digium phones.

2012-06-04 20:26:12 +00:00

chan_skinny.c

Various small chan_skinny fixes and cleanup

2012-06-15 21:18:56 +00:00

chan_unistim.c

Fix MWI update so LED display correct voicemail state after phone usage. Also fixes few warnings.

2012-06-08 08:32:49 +00:00

chan_vpb.cc

Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)

2012-05-14 19:44:27 +00:00

console_board.c

Multiple revisions 369001-369002

2012-06-15 16:20:16 +00:00

console_gui.c

Multiple revisions 369001-369002

2012-06-15 16:20:16 +00:00

console_video.c

Multiple revisions 369001-369002

2012-06-15 16:20:16 +00:00

console_video.h

implement the 'freeze' function for incoming frames;

2008-06-30 15:45:15 +00:00

iax2-parser.c

Multiple revisions 369001-369002

2012-06-15 16:20:16 +00:00

iax2-parser.h

Asterisk media architecture conversion - no more format bitfields

2011-02-03 16:22:10 +00:00

iax2-provision.c

Multiple revisions 369001-369002

2012-06-15 16:20:16 +00:00

iax2-provision.h

Don't reload a configuration file if nothing has changed.

2007-08-16 21:09:46 +00:00

iax2.h

Media Project Phase2: SILK 8khz-24khz, SLINEAR 8khz-192khz, SPEEX 32khz, hd audio ConfBridge, and other stuff

2011-02-22 23:04:49 +00:00

Makefile

Extract sig_ss7 out of chan_dahdi.

2010-06-07 20:04:42 +00:00

misdn_config.c

Multiple revisions 369001-369002

2012-06-15 16:20:16 +00:00

sig_analog.c

Unique Call ID logging Phases III and IV

2012-06-26 21:45:22 +00:00

sig_analog.h

push 'outgoing' flag from sig_XXX up to chan_dahdi

2012-02-18 08:02:08 +00:00

sig_pri.c

Unique Call ID logging Phases III and IV

2012-06-26 21:45:22 +00:00

sig_pri.h

Clear ISDN channel resetting state if the peer continues to use it.

2012-04-25 19:55:12 +00:00

sig_ss7.c

Unique Call ID logging Phases III and IV

2012-06-26 21:45:22 +00:00

sig_ss7.h

Setup DSP when SS7 call is connected or early media is available.

2012-03-05 21:48:32 +00:00

vcodecs.c

Multiple revisions 369001-369002

2012-06-15 16:20:16 +00:00

vgrabbers.c

Multiple revisions 369001-369002

2012-06-15 16:20:16 +00:00