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Michael Kuron d73f22268f res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip ... chan_sip supported sending AOC-D and AOC-E information in SIP INFO messages in an "AOC" header in a format that was originally defined by Snom. In the meantime, ETSI TS 124 647 introduced an XML-based AOC format that is supported by devices from multiple vendors, including Snom phones with firmware >= 8.4.2 (released in 2010). This commit adds a new res_pjsip_aoc module that inserts AOC information into outgoing messages or sends SIP INFO messages as described below. It also fixes a small issue in res_pjsip_session which didn't always call session supplements on outgoing_response. * AOC-S in the 180/183/200 responses to an INVITE request * AOC-S in SIP INFO (if a 200 response has already been sent or if the INVITE was sent by Asterisk) * AOC-D in SIP INFO * AOC-D in the 200 response to a BYE request (if the client hangs up) * AOC-D in a BYE request (if Asterisk hangs up) * AOC-E in the 200 response to a BYE request (if the client hangs up) * AOC-E in a BYE request (if Asterisk hangs up) The specification defines one more, AOC-S in an INVITE request, which is not implemented here because it is not currently possible in Asterisk to have AOC data ready at this point in call setup. Once specifying AOC-S via the dialplan or passing it through from another SIP channel's INVITE is possible, that might be added. The SIP INFO requests are sent out immediately when the AOC indication is received. The others are inserted into an appropriate outgoing message whenever that is ready to be sent. In the latter case, the XML is stored in a channel variable at the time the AOC indication is received. Depending on where the AOC indications are coming from (e.g. PRI or AMI), it may not always be possible to guarantee that the AOC-E is available in time for the BYE. Successfully tested AOC-D and both variants of AOC-E with a Snom D735 running firmware 10.1.127.10. It does not appear to properly support AOC-S however, so that could only be tested by inspecting SIP traces. ASTERISK-21502 #close Reported-by: Matt Jordan <mjordan@digium.com> Change-Id: Iebb7ad0d5f88526bc6629d3a1f9f11665434d333		2 years ago
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include	res_pjsip: Fix for Doxygen.	3 years ago
config_auth.c	res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability	6 years ago
config_domain_aliases.c	res_pjsip: Make pjlib.h consistently included.	6 years ago
config_global.c	res_pjsip: return all codecs on a re-INVITE without SDP	2 years ago
config_system.c	res_pjsip.c: Added disable_rport option for pjsip.conf	4 years ago
config_transport.c	pjsip: Add TLS transport reload support for certificate and key.	2 years ago
location.c	time: add support for time64 libcs	3 years ago
pjsip_cli.c	pjsip: Improve CLI completion performance	7 years ago
pjsip_config.xml	res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip	2 years ago
pjsip_configuration.c	res_pjsip_aoc: New module for sending advice-of-charge with chan_pjsip	2 years ago
pjsip_distributor.c	res_pjsip: Add TEL URI support for basic calls.	2 years ago
pjsip_global_headers.c	Remove redundant module checks and references.	7 years ago
pjsip_manager.xml	xmldoc: Fix issue with xmlstarlet validation	3 years ago
pjsip_message_filter.c	res_pjsip: Add TEL URI support for basic calls.	2 years ago
pjsip_options.c	res_pjsip: Add mediasec capabilities.	2 years ago
pjsip_resolver.c	res: Fix for Doxygen.	3 years ago
pjsip_scheduler.c	res: Spelling fixes	3 years ago
pjsip_session.c	res: Spelling fixes	3 years ago
pjsip_transport_events.c	pjsip_transport_events: Fix possible use after free on transport	2 years ago
pjsip_transport_management.c	res_pjsip, res_pjsip_session: initialize local variables	4 years ago
presence_xml.c	res_pjsip_pidf_eyebeam_body_supplement: Correct status presentation	7 years ago
security_agreements.c	res_pjsip: Add mediasec capabilities.	2 years ago
security_events.c	security-events: Fix SuccessfulAuth using_password declaration.	7 years ago