asterisk/res/res_pjsip.exports.in

{
	global:
		LINKER_SYMBOL_PREFIXast_sip_*;
		LINKER_SYMBOL_PREFIX__ast_sip_*;
		LINKER_SYMBOL_PREFIXast_copy_pj_str;
		LINKER_SYMBOL_PREFIXast_copy_pj_str2;
		LINKER_SYMBOL_PREFIXast_pjsip_rdata_get_endpoint;
	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:`
res_pjsip: AMI commands and events. Created the following AMI commands and corresponding events for res_pjsip: PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few select attributes on each. Events: EndpointList - for each endpoint a few attributes. EndpointlistComplete - after all endpoints have been listed. PJSIPShowEndpoint - Provides a detail list of attributes for a specified endpoint. Events: EndpointDetail - attributes on an endpoint. AorDetail - raised for each AOR on an endpoint. AuthDetail - raised for each associated inbound and outbound auth TransportDetail - transport attributes. IdentifyDetail - attributes for the identify object associated with the endpoint. EndpointDetailComplete - last event raised after all detail events. PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound registrations. Events: InboundRegistrationDetail - inbound registration attributes for each registration. InboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound registrations. Events: OutboundRegistrationDetail - outbound registration attributes for each registration. OutboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. (issue ASTERISK-22609) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2013-11-23 17:26:57 +00:00			`LINKER_SYMBOL_PREFIXast_sip_*;`
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_*;`
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			`LINKER_SYMBOL_PREFIXast_copy_pj_str;`
res_pjsip: Add "webrtc" configuration option This patch creates a new configuration option called "webrtc". When enabled it defaults and enables the following options that are needed in order for webrtc to work in Asterisk: rtcp-mux, use_avpf, ice_support, and use_received_transport=enabled media_encryption=dtls dtls_verify=fingerprint dtls_setup=actpass When "webrtc" is enabled, this patch also parses the "msid" media level attribute from an SDP. It will also appropriately add it onto the outgoing session when applicable. Lastly, when "webrtc" is enabled h264 RTCP FIR feedback frames are now sent. ASTERISK-27119 #close Change-Id: I5ec02e07c5d5b9ad86a34fdf31bf2f9da9aac6fd 2017-07-10 18:17:44 -05:00			`LINKER_SYMBOL_PREFIXast_copy_pj_str2;`
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			`LINKER_SYMBOL_PREFIXast_pjsip_rdata_get_endpoint;`
			`local:`
			`*;`
			`};`