30023 Commits (e82b921c359bcd359e6f632a7ef128c3de8d84ec)

 

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Author	SHA1	Message	Date
Corey Farrell	e82b921c35	Modules: Fix issues with CLI completion. ... * Stop using ast_module_helper to check if a module is loaded, use ast_module_check instead (app_confbridge and app_meetme). * Stop ast_module_helper from listing reload classes when needsreload was not requested. ASTERISK-27378 Change-Id: Iaed8c1e4fcbeb242921dbac7929a0fe75ff4b239	8 years ago
Joshua Colp	2a898460db	Merge "codec.c: Defensively check the returned samples."	8 years ago
Joshua Colp	2fc5b92d59	Merge "app_voicemail.c: Fix compiler warning with IMAP build."	8 years ago
Joshua Colp	99f998c69d	Merge "res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint."	8 years ago
Richard Mudgett	26607e4e3b	app_voicemail.c: Fix compiler warning with IMAP build. ... ASTERISK-27181 Change-Id: Ic4468b49860bd7f67e922baf4c9e96828c184d17	8 years ago
Richard Mudgett	2ca3dbb197	codec.c: Defensively check the returned samples. ... Earlier versions of the codec_opus samples_count callback can return negative error values on undecodable frames. This resulted in a divide by zero exception. * Added a defensive check in ast_codec_samples_count() for a "negative" samples count return value. Log the event and set the count to zero. ASTERISK-27194 Change-Id: Icf69350307ecbbc80a3d74de46af9bd80ea17819	8 years ago
Joshua Colp	4f2717bab0	Merge "Single API for ast_store_lock_info and ast_remove_lock_info."	8 years ago
Joshua Colp	9e1fbab382	res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint. ... When the identify_by option on an endpoint is set to ip it will only be identified using the res_pjsip_endpoint_identifier_ip module. This ensures that it is not mistakenly matched using the username of the From header. To ensure behavior has not changed the default has been changed to "username,ip" for the identify_by option. ASTERISK-27206 Change-Id: I2170b86a7f7e221b4f00bf14aa1ef1ac5b050bbd	8 years ago
Ben Ford	3821be1c68	http.c: Fix http header send content. ... Currently ast_http_send barricades a portion of the content that needs to be sent in order to establish a connection for things like the ARI client. The conditional and contents have been changed to ensure that everything that needs to be sent, will be sent. ASTERISK-27372 Change-Id: I8816d2d8f80f4fefc6dcae4b5fdfc97f1e46496d	8 years ago
Joshua Colp	65357091d8	Merge "Bundled pjproject: Enable pj_assert when dev-mode is enabled."	8 years ago
Joshua Colp	0436fe18f0	Merge "chan_sip: Fix SUBSCRIBE with missing "Expires" header."	8 years ago
Joshua Colp	61aeeb6d52	Merge "hashtab: Use ast_free."	8 years ago
Joshua Colp	2f2bbbd43f	Merge "res_pjsip_sdp_rtp: Fix setting of address type for rtp_ipv6"	8 years ago
Joshua Colp	3333e787ab	Merge "main/Makefile: Remove rule for non-existant testexpr2."	8 years ago
Joshua Colp	0bbd26e49d	Merge "res_xmpp: Ensure the connection filter is available."	8 years ago
Corey Farrell	569e9a8391	Single API for ast_store_lock_info and ast_remove_lock_info. ... This makes the 'bt' parameter unconditional for ast_store_lock_info and ast_remove_lock_info. The 'bt' parameter is unused when HAVE_BKTR is undefined. Change-Id: Ieced0e920928b735a39c3b5952b806c473d67453	8 years ago
George Joseph	0b05dafc89	Merge "lpc10: Avoid compiler warning when DONT_OPTIMIZE/COMPILE_DOUBLE."	8 years ago
Joshua Colp	909fe2f833	Merge "test_config: Fix failure and segfault when config_hook is run twice."	8 years ago
Corey Farrell	6474de5f72	chan_sip: Fix SUBSCRIBE with missing "Expires" header. ... When chan_sip receives a SUBSCRIBE request with no "Expires" header it processes the request as an unsubscribe. This is incorrect, per RFC3264 when the "Expires" header is missing a default expiry should be used. ASTERISK-18140 Change-Id: Ibf6dcd4fdd07a32c2bc38be1dd557981f08188b5	8 years ago
Joshua Colp	4cc36594ef	Merge "res_pjproject.c: Upgrade bundled PJPROJECT to 2.7"	8 years ago
Joshua Colp	2842ae0daf	Merge "chan_sip: Crypto attribute not last but first on SDP media level."	8 years ago
Alexander Traud	7126520b3e	lpc10: Avoid compiler warning when DONT_OPTIMIZE/COMPILE_DOUBLE. ... ASTERISK-23556 Reported by: Marcello Ceschia Change-Id: Ic27e88e0336a0d83877dc857938659dc5560b93c	8 years ago
Corey Farrell	841ac3ded6	hashtab: Use ast_free. ... A few places in hashtab use free instead of ast_free, remove declaration of ASTMM_LIBC from hashtab.c as it's no longer needed. Change-Id: I2ff089bad71640c03c3ce97f1b00fc962ef79427	8 years ago
Corey Farrell	fb585cf185	Bundled pjproject: Enable pj_assert when dev-mode is enabled. ... ASTERISK-27359 Change-Id: Ib01fb6c01f9bb87129374a51cb9318c474147517	8 years ago
Corey Farrell	ee21076151	main/Makefile: Remove rule for non-existant testexpr2. ... Change-Id: Ibb3e47f27a395d74d8c5263db015b05434f5969b	8 years ago
Corey Farrell	a9e9608982	test_config: Fix failure and segfault when config_hook is run twice. ... On second run the config_hook test was unexpectedly failing to load test_config.conf because it was still unmodified since the last load. This is fixed by not passing CONFIG_FLAG_FILEUNCHANGED for the initial loads, only using it when we are tested that a reload of unmodified files do not initiate the hook. ASTERISK-25960 Change-Id: Ifd679509a23ed163e5cc647490bf7df4ae3cd856	8 years ago
George Joseph	6f0431798e	res_pjsip_sdp_rtp: Fix setting of address type for rtp_ipv6 ... create_outgoing_sdp_stream was setting "addr_type = STR_IP6" only when an ipv6 media_address was specified on the endpoint. If rtp_ipv6 was set and ast_sip_get_host_ip_string returned an ipv6 address, we were leaving the addr_type set at the default of STR_IP4. This caused the address type to be set incorrectly on the "o" and "c" SDP attributes even though the address was set correctly. Some clients don't like the mismatch. * Removed the test for endpoint/media_address and now check all addresses for ipv6. ASTERISK-27198 Reported by: Martin Cisárik Change-Id: I5214fc31b728117842243807e7927a319cf77592	8 years ago
Joshua Colp	8b719a3e48	Merge "bridge_softmix: Reduce topology cloning and improve renegotiation."	8 years ago
Richard Mudgett	488f98310f	app_agent_pool.c: Fix online documentation typo. ... Change-Id: Ib0bc95fd0ec288c78c313823254d7a84ebfc4429	8 years ago
Joshua Colp	252353e0a9	res_xmpp: Ensure the connection filter is available. ... Users of the API that res_xmpp provides expect that a filter be available on the client at all times. When OAuth authentication support was added this requirement was not maintained. This change merely moves the OAuth authentication to after the filter is created, ensuring users of res_xmpp can add things to the filter as needed. ASTERISK-27346 Change-Id: I4ac474afe220e833288ff574e32e2b9a23394886	8 years ago
Alexander Traud	840e08716b	chan_sip: Crypto attribute not last but first on SDP media level. ... This matches the behavior of the other SIP channel driver, chan_pjsip. ASTERISK-27365 Change-Id: I8f23a51290a58b75816da2999ed1965441dfc5d6	8 years ago
Richard Mudgett	e41561fc2a	res_pjproject.c: Upgrade bundled PJPROJECT to 2.7 ... Update patches included in bundled PJPROJECT for the new version. ASTERISK-27355 Change-Id: I9ac5dbbffaadca25ad24fac8b9ab615e5ace6083	8 years ago
Corey Farrell	4760b2445c	res_pjsip_pubsub: Prevent unload except during shutdown. ... Prevent unload of the module as certain pjsip initialization functions cannot be reversed. This required a reorder of the module_load so that the non-reversable pjsip functions are not called until all potential errors have been ruled out. ASTERISK-24483 Change-Id: Iee900f20bdd6ee1bfe23efdec0d87765eadce8a7	8 years ago
Joshua Colp	6bb92a7164	Merge "chan_sip: Fix output of 'sip set debug off'."	8 years ago
Joshua Colp	eb59d80782	Merge "res_pjsip_refer: Prevent unload except during shutdown."	8 years ago
Joshua Colp	7b8f199a50	Merge "app_macro deprecation."	8 years ago
Corey Farrell	449ee66a11	res_pjsip_refer: Prevent unload except during shutdown. ... Prevent unload of the module as certain pjsip initialization functions cannot be reversed. ASTERISK-24483 Change-Id: I94597ec8b8491f5af9c57bf66dbc3b078fe2d49d	8 years ago
Corey Farrell	c9e19b31f5	chan_sip: Fix output of 'sip set debug off'. ... When sip.conf contains 'sipdebug=yes' it is impossible to disable it using CLI 'sip set debug off'. This corrects the output of that CLI command to instruct the user to turn sipdebug off in the configuration file. ASTERISK-23462 #close Change-Id: I1cceade9caa9578e1b060feb832e3495ef5ad318	8 years ago
Corey Farrell	955a891a84	app_macro deprecation. ... * Mark the module deprecated. * Disable the module by default. * Produce a warning the first time a macro is used. * Note deprecation related options in app_dial and app_queue. ASTERISK-27350 Change-Id: I560ea043bacdbc5534a17d97854273d52c2f1bdc	8 years ago
Alexander Traud	95b45d1c46	res_srtp: Add support for libsrtp2 with AES-GCM. ... Beside allowing AES-GCM again, this adds AES-192 again. ASTERISK-27356 Change-Id: Ia97a435faf26300335d9552fa676b5d17e5f7233	8 years ago
Jenkins2	214a40b6b5	Merge "res_corosync: Fix linking issue with Corosync 2.x"	8 years ago
Joshua Colp	206b66db55	Merge "cdr.c: Rename the Party A CDR container."	8 years ago
Jenkins2	203c8d6fa1	Merge "cdr.c: Add container to key off of Party B channel names."	8 years ago
Jenkins2	8495b373ad	Merge "contrib/script/sip_to_pjsip: implement 'all' for allow/disallow"	8 years ago
Jenkins2	3f6e3221e3	Merge "bridge_simple: Improve renegotiation success rate."	8 years ago
Jenkins2	027ad49b08	Merge "chan_dahdi: wrap include file which is not present on BSD systems in #ifdef"	8 years ago
Jenkins2	c5d9cce6f6	Merge "res_pjsip_session: Rewrite o= with external_media_address."	8 years ago
Joshua Colp	5d8c517960	bridge_softmix: Reduce topology cloning and improve renegotiation. ... As channels join and leave an SFU the bridge_softmix module needs to renegotiate to add and remove their streams from the other participants. Previously this was done by constructing the ideal stream topology every time but in the case of leave this was incomplete. This change makes it so bridge_softmix keeps an ideal stream topology for each channel and uses it when making changes. This ensures that when we request a renegotiation we are always certain that we are aiming for the best stream topology possible. In the case of a channel leaving this ensures that we try to have an existing participant fill their place if a participant has a fixed limit on the maximum number of video streams they allow. ASTERISK-27354 Change-Id: I58070f421ddeadd2844a33b869b052630cf2e514	8 years ago
Richard Mudgett	73164d0d7f	cdr.c: Rename the Party A CDR container. ... * Rename the Party A CDR container from active_cdrs_by_channel to active_cdrs_master. * Renamed the support functions associated with active_cdrs_master appropriately. ASTERISK-27335 Change-Id: I6104bb3edc3a0b7243ce502e45e8832b0cff14f7	8 years ago
Richard Mudgett	fe1120cf88	cdr.c: Add container to key off of Party B channel names. ... The CDR performance gets worse the further it gets behind in processing stasis messages. One of the reasons is because of a n*m loop used when processing Party B information. * Added a new CDR container that is keyed to Party B so we don't need such a large loop when processing Party B information. NOTE: To reduce the size of the patch I deferred to another patch the renaming of the Party A active_cdrs_by_channel container to active_cdrs_master and renaming the container's hash and cmp functions appropriately. ASTERISK-27335 Change-Id: I0bf66e8868f8adaa4b5dcf9e682e34951c350249	8 years ago