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2017-10-25Merge "res_pjsip_sdp_rtp: Fix setting of address type for rtp_ipv6"Joshua Colp
2017-10-25Merge "main/Makefile: Remove rule for non-existant testexpr2."Joshua Colp
2017-10-24Merge "res_xmpp: Ensure the connection filter is available."Joshua Colp
2017-10-24Single API for ast_store_lock_info and ast_remove_lock_info.Corey Farrell
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
2017-10-24Merge "lpc10: Avoid compiler warning when DONT_OPTIMIZE/COMPILE_DOUBLE."George Joseph
2017-10-24Merge "test_config: Fix failure and segfault when config_hook is run twice."Joshua Colp
2017-10-24chan_sip: Fix SUBSCRIBE with missing "Expires" header.Corey Farrell
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
2017-10-24Merge "res_pjproject.c: Upgrade bundled PJPROJECT to 2.7"Joshua Colp
2017-10-24Merge "chan_sip: Crypto attribute not last but first on SDP media level."Joshua Colp
2017-10-24lpc10: Avoid compiler warning when DONT_OPTIMIZE/COMPILE_DOUBLE.Alexander Traud
ASTERISK-23556 Reported by: Marcello Ceschia Change-Id: Ic27e88e0336a0d83877dc857938659dc5560b93c
2017-10-23hashtab: Use ast_free.Corey Farrell
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
2017-10-23Bundled pjproject: Enable pj_assert when dev-mode is enabled.Corey Farrell
ASTERISK-27359 Change-Id: Ib01fb6c01f9bb87129374a51cb9318c474147517
2017-10-23main/Makefile: Remove rule for non-existant testexpr2.Corey Farrell
Change-Id: Ibb3e47f27a395d74d8c5263db015b05434f5969b
2017-10-23test_config: Fix failure and segfault when config_hook is run twice.Corey Farrell
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
2017-10-23res_pjsip_sdp_rtp: Fix setting of address type for rtp_ipv6George Joseph
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
2017-10-23Merge "bridge_softmix: Reduce topology cloning and improve renegotiation."Joshua Colp
2017-10-23app_agent_pool.c: Fix online documentation typo.Richard Mudgett
Change-Id: Ib0bc95fd0ec288c78c313823254d7a84ebfc4429
2017-10-22res_xmpp: Ensure the connection filter is available.Joshua Colp
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
2017-10-21chan_sip: Crypto attribute not last but first on SDP media level.Alexander Traud
This matches the behavior of the other SIP channel driver, chan_pjsip. ASTERISK-27365 Change-Id: I8f23a51290a58b75816da2999ed1965441dfc5d6
2017-10-20res_pjproject.c: Upgrade bundled PJPROJECT to 2.7Richard Mudgett
Update patches included in bundled PJPROJECT for the new version. ASTERISK-27355 Change-Id: I9ac5dbbffaadca25ad24fac8b9ab615e5ace6083
2017-10-19res_pjsip_pubsub: Prevent unload except during shutdown.Corey Farrell
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
2017-10-19Merge "chan_sip: Fix output of 'sip set debug off'."Joshua Colp
2017-10-19Merge "res_pjsip_refer: Prevent unload except during shutdown."Joshua Colp
2017-10-18Merge "app_macro deprecation."Joshua Colp
2017-10-18res_pjsip_refer: Prevent unload except during shutdown.Corey Farrell
Prevent unload of the module as certain pjsip initialization functions cannot be reversed. ASTERISK-24483 Change-Id: I94597ec8b8491f5af9c57bf66dbc3b078fe2d49d
2017-10-18chan_sip: Fix output of 'sip set debug off'.Corey Farrell
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
2017-10-18app_macro deprecation.Corey Farrell
* 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
2017-10-18res_srtp: Add support for libsrtp2 with AES-GCM.Alexander Traud
Beside allowing AES-GCM again, this adds AES-192 again. ASTERISK-27356 Change-Id: Ia97a435faf26300335d9552fa676b5d17e5f7233
2017-10-17Merge "res_corosync: Fix linking issue with Corosync 2.x"Jenkins2
2017-10-17Merge "cdr.c: Rename the Party A CDR container."Joshua Colp
2017-10-17Merge "cdr.c: Add container to key off of Party B channel names."Jenkins2
2017-10-17Merge "contrib/script/sip_to_pjsip: implement 'all' for allow/disallow"Jenkins2
2017-10-17Merge "bridge_simple: Improve renegotiation success rate."Jenkins2
2017-10-17Merge "chan_dahdi: wrap include file which is not present on BSD systems in ↵Jenkins2
#ifdef"
2017-10-17Merge "res_pjsip_session: Rewrite o= with external_media_address."Jenkins2
2017-10-17bridge_softmix: Reduce topology cloning and improve renegotiation.Joshua Colp
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
2017-10-16cdr.c: Rename the Party A CDR container.Richard Mudgett
* 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
2017-10-16cdr.c: Add container to key off of Party B channel names.Richard Mudgett
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
2017-10-16Merge "declare optional openssl dependencies in moduleinfo"Jenkins2
2017-10-16contrib/script/sip_to_pjsip: implement 'all' for allow/disallowTorrey Searle
when 'all' is specified in an allow or disallow section, it should erase all values from the inverse section in the default config. E.G. allow=all should erase any deny values from default config & vice-versa ASTERISK-27333 #close Change-Id: I99219478fb98f08751d769daaee0b7795118a5a6
2017-10-16Merge "cdr.c: Eliminated many calls to ao2_global_obj_ref()."Joshua Colp
2017-10-16Merge "cdr.c: Defer getting ao2_global_obj_ref() until needed."Joshua Colp
2017-10-15Merge "ast_bt_get_symbols: Prevent double-free."Jenkins2
2017-10-15Merge "tcptls: NULL-check the parameter of ast_ssl_teardown before accessing ↵Joshua Colp
it."
2017-10-14chan_dahdi: wrap include file which is not present on BSD systems in #ifdefGuido Falsi
The sys/sysmacros.h include file does not exist in BSD systems and is not required to build this module there. Since an "#if defined(__NetBSD__) || defined(__FreeBSD__)" section already exist I moved that include line inside it's #else branch. ASTERISK-27343 #close Change-Id: Ibfb64f4e9a0ce8b6eda7a7695cfe57916f175dc1
2017-10-14res_pjsip_session: Rewrite o= with external_media_address.Alexander Traud
PJSIP allows a domain name as external_media_address. This allows chan_pjsip to be used behind a NAT with changing IP addresses. The IP address of that domain is resolved to the c= line already. This change sets also the o= line to that domain. ASTERISK-27341 #close Change-Id: I690163b6e762042ec38b3995aa5c9bea909d8ec4
2017-10-13Merge "cdr.c: Set stringfields only if they are different."Jenkins2
2017-10-13bridge_simple: Improve renegotiation success rate.Joshua Colp
When making channels compatible the bridge_simple module will renegotiate one to better match the other. Some endpoints incorrectly terminate the call if this process fails. To better handle this scenario the audio streams present on the new requested topology will include any existing negotiated formats that happen to exist on the first valid audio stream. This ensures formats are persent that are known to be acceptable to the remote endpoint. ASTERISK-27259 Change-Id: I8fc0cc03e8bcfd0be8302f13b9f32d8268977f43
2017-10-13Merge "cdr.c: Fix setting dnid, callingsubaddr, and calledsubaddr"Kevin Harwell
2017-10-13ast_bt_get_symbols: Prevent double-free.Corey Farrell
It's possible for bfdobj to be created but syms not created. If syms was not allocated in the current loop iteration but was allocated in the previous iteration it would crash. ASTERISK-27340 Change-Id: I5b110c609f6dfe91339f782a99a431bca5837363