asterisk/res/res_pjsip_session.exports.in

{
	global:
		LINKER_SYMBOL_PREFIXast_sip_session_*;
		LINKER_SYMBOL_PREFIX__ast_sip_session_*;
		LINKER_SYMBOL_PREFIXast_sip_dialog_get_session;
		LINKER_SYMBOL_PREFIXast_sip_channel_pvt_alloc;
		LINKER_SYMBOL_PREFIXast_sip_create_joint_call_cap;
	local:
		*;
};
Merge the pimp_my_sip branch into trunk. The pimp_my_sip branch is being merged at this point because it offers basic functionality, and from an API standpoint, things are complete. SIP work is not feature-complete; however, with the completion of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have been created, and thus it is possible for developers to attempt to create new SIP work. API documentation can be found in the doxygen in the code, but usability documentation is still lacking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2013-04-25 18:25:31 +00:00			`{`
			`global:`
chan_pjsip: Add support for multiple streams of the same type. The stream topology (list of streams and order) is now stored with the configured PJSIP endpoints and used during the negotiation process. Media negotiation state information has been changed to be stored in a separate object. Two of these objects exist at any one time on a session. The active media state information is what was previously negotiated and the pending media state information is what the media state will become if negotiation succeeds. Streams and other state information is stored in this object using the index (or position) of each individual stream for easy lookup. The ability for a media type handler to specify a callback for writing has been added as well as the ability to add file descriptors with a callback which is invoked when data is available to be read on them. This allows media logic to live outside of the chan_pjsip module. Direct media has been changed so that only the first audio and video stream are directly connected. In the future once the RTP engine glue API has been updated to know about streams each individual stream can be directly connected as appropriate. Media negotiation itself will currently answer all the provided streams on an offer within configured limits and on an offer will use the topology created as a result of the disallow/allow codec lines. If a stream has been removed or declined we will now mark it as such within the resulting SDP. Applications can now also request that the stream topology change. If we are told to do so we will limit any provided formats to the ones configured on the endpoint and send a re-invite with the new topology. Two new configuration options have also been added to PJSIP endpoints: max_audio_streams: determines the maximum number of audio streams to offer/accept from an endpoint. Defaults to 1. max_video_streams: determines the maximum number of video streams to offer/accept from an endpoint. Defaults to 1. ASTERISK-27076 Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7 2017-05-30 09:12:47 -05:00			`LINKER_SYMBOL_PREFIXast_sip_session_*;`
res_pjsip: Add REF_DEBUG info to module references. This provides better information to REF_DEBUG log for troubleshooting when the system is unable to unload res_pjsip.so during shutdown due to module references. ASTERISK-27306 Change-Id: I63197ad33d1aebe60d12e0a6561718bdc54e4612 2017-10-04 11:59:49 -04:00			`LINKER_SYMBOL_PREFIX__ast_sip_session_*;`
Merge in current pimp_my_sip work, including: 1. Security events 2. Websocket support 3. Diversion header + redirecting support 4. An anonymous endpoint identifier 5. Inbound extension state subscription support 6. PIDF notify generation 7. One touch recording support (special thanks Sean Bright!) 8. Blind and attended transfer support 9. Automatic inbound registration expiration 10. SRTP support 11. Media offer control dialplan function 12. Connected line support 13. SendText() support 14. Qualify support 15. Inband DTMF detection 16. Call and pickup groups 17. Messaging support Thanks everyone! Side note: I'm reminded of the song "How Far We've Come" by Matchbox Twenty. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2013-06-22 14:03:22 +00:00			`LINKER_SYMBOL_PREFIXast_sip_dialog_get_session;`
Expose the chan_pjsip implementation pvt and session in a defined manner. This allows modules outside of chan_pjsip itself to get the session given only an Asterisk channel. Review: https://reviewboard.asterisk.org/r/2674/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2013-07-23 12:27:03 +00:00			`LINKER_SYMBOL_PREFIXast_sip_channel_pvt_alloc;`
test_res_pjsip_session_caps: Create unit test This unit test runs through combinations of... * Local codecs * Remote Codecs * Codec Preference * Incoming/Outgoing A few new APIs were created to make it easier to test the functionality but didn't result in any actual functional change. ASTERISK_28777 Change-Id: Ic8957c43e7ceeab0e9272af60ea53f056164f164 2020-03-25 11:51:32 -06:00			`LINKER_SYMBOL_PREFIXast_sip_create_joint_call_cap;`
Merge the pimp_my_sip branch into trunk. The pimp_my_sip branch is being merged at this point because it offers basic functionality, and from an API standpoint, things are complete. SIP work is not feature-complete; however, with the completion of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have been created, and thus it is possible for developers to attempt to create new SIP work. API documentation can be found in the doxygen in the code, but usability documentation is still lacking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2013-04-25 18:25:31 +00:00			`local:`
			`*;`
			`};`