[Asterisk-code-review] res pjsip refer/session: Calls dropped during transfer (asterisk[certified/13.13])

[Kevin Harwell]

Hello Jenkins2, 

I'd like you to reexamine a change. Please visit

    https://gerrit.asterisk.org/5821

to look at the new patch set (#2).

Change subject: res_pjsip_refer/session: Calls dropped during transfer
......................................................................

res_pjsip_refer/session: Calls dropped during transfer

When doing an attended transfer it's possible for the transferer, after
receiving an accepted response from Asterisk, to send a BYE to Asterisk,
which can then be processed before Asterisk has time to start and/or
complete the transfer process. This of course causes the transfer to not
complete successfully, thus dropping the call.

This patch makes it so any BYEs received from the transferer, after the REFER,
that initiate a session end are deferred until the transfer is complete. This
allows the channel that would have otherwise been hung up by Asterisk to
remain available throughout the transfer process.

ASTERISK-27053 #close

Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a
---
M include/asterisk/res_pjsip_session.h
M res/res_pjsip_refer.c
M res/res_pjsip_session.c
M res/res_pjsip_session.exports.in
4 files changed, 48 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/21/5821/2
-- 
To view, visit https://gerrit.asterisk.org/5821
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Gerrit-Project: asterisk
Gerrit-Branch: certified/13.13
Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a
Gerrit-Change-Number: 5821
Gerrit-PatchSet: 2
Gerrit-Owner: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Jenkins2
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