Add a bunch of options from sip.conf to res_sip.conf
authorMark Michelson <mmichelson@digium.com>
Thu, 18 Jul 2013 19:25:51 +0000 (19:25 +0000)
committerMark Michelson <mmichelson@digium.com>
Thu, 18 Jul 2013 19:25:51 +0000 (19:25 +0000)
commitc47787feab34d9572b41ebe7148c169b52362fbd
tree85410d3780938b31d9605f6d05cef9a744c3932d
parent3c86832f9f271c8d479cf956155424fda512c76b
Add a bunch of options from sip.conf to res_sip.conf

For a complete list of the options added, see the review linked
at the bottom of this commit message.

(closes issue ASTERISK-21506)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2671

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394759 65c4cc65-6c06-0410-ace0-fbb531ad65f3
17 files changed:
channels/chan_gulp.c
include/asterisk/res_sip.h
res/res_sip.c
res/res_sip.exports.in
res/res_sip/config_global.c [new file with mode: 0644]
res/res_sip/config_system.c [new file with mode: 0644]
res/res_sip/config_transport.c
res/res_sip/include/res_sip_private.h
res/res_sip/sip_configuration.c
res/res_sip/sip_global_headers.c [new file with mode: 0644]
res/res_sip_caller_id.c
res/res_sip_mwi.c
res/res_sip_one_touch_record_info.c
res/res_sip_pubsub.c
res/res_sip_refer.c
res/res_sip_sdp_rtp.c
res/res_sip_session.c