]> git.ipfire.org Git - thirdparty/asterisk.git/commit
ari/pjsip: Make it possible to control transfers through ARI
authorHolger Hans Peter Freyther <holger@moiji-mobile.com>
Sat, 15 Jun 2024 08:01:58 +0000 (16:01 +0800)
committerAsterisk Development Team <asteriskteam@digium.com>
Thu, 20 Mar 2025 18:18:29 +0000 (18:18 +0000)
commit6f17f61394c8693d0fbb50bf6943c0ebacecf787
tree044a0d141dc31582b17c369cc8c7ea89792a5c77
parentaf966a98a16b7410f7257cd4eae549660f6c804a
ari/pjsip: Make it possible to control transfers through ARI

Introduce a ChannelTransfer event and the ability to notify progress to
ARI. Implement emitting this event from the PJSIP channel instead of
handling the transfer in Asterisk when configured.

Introduce a dialplan function to the PJSIP channel to switch between the
"core" and "ari-only" behavior.

UserNote: Call transfers on the PJSIP channel can now be controlled by
ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only)
dialplan function.

(cherry picked from commit 5e4fca062c9f4b4f0aaf5ce9572ae5a3379d6b25)
18 files changed:
channels/chan_pjsip.c
channels/pjsip/dialplan_functions.c
channels/pjsip/dialplan_functions_doc.xml
channels/pjsip/include/dialplan_functions.h
include/asterisk/frame.h
include/asterisk/refer.h
include/asterisk/res_pjsip_session.h
include/asterisk/stasis_channels.h
main/refer.c
main/stasis_channels.c
res/ari/ari_model_validators.c
res/ari/ari_model_validators.h
res/ari/resource_channels.c
res/ari/resource_channels.h
res/res_ari_channels.c
res/res_pjsip_refer.c
rest-api/api-docs/channels.json
rest-api/api-docs/events.json