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2017-10-27Merge "app_voicemail.c: Fix compiler warning with IMAP build." into 15Jenkins2
2017-10-26Merge "res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint." ↵Jenkins2
into 15
2017-10-26app_voicemail.c: Fix compiler warning with IMAP build.Richard Mudgett
ASTERISK-27181 Change-Id: Ic4468b49860bd7f67e922baf4c9e96828c184d17
2017-10-26codec.c: Defensively check the returned samples.Richard Mudgett
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
2017-10-25Merge "http.c: Fix http header send content." into 15George Joseph
2017-10-25res_pjsip: Add 'ip' as a valid option to 'identify_by' on endpoint.Joshua Colp
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
2017-10-25ast_coredumper: Add gzipping of binaries and display of signal infoGeorge Joseph
The --tarball-coredump option now creates a gzipped tarball of coredumps processed, their results txt files and copies of /etc/os-release, /usr/sbin/asterisk, /usr/lib(64)/libasterisk* and /usr/lib(64)/asterisk as those files are needed to properly examine the coredump. The file will be named /tmp/asterisk.<timestamp>.coredumps.tar.gz or /tmp/asterisk-<uniqueid>.coredumps.tar.gz if --tarball-uniqueid was specified. Added dumps of *_siginfo to the top of the txt files so you can tell what signal was invoked. Change-Id: Ib9ee6d83592d4b1bc90cb3419a05376a88d1ded9
2017-10-25http.c: Fix http header send content.Ben Ford
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
2017-10-25Build System: Fix --disable-xmldoc option.Corey Farrell
The configure option to disable XML documentation does not currently work. This patch makes it effective, but also causes an ABI change by removing the ast_xmldoc_* symbols. Disabling xmldoc also prevents docs from being automatically generated, but they can still be manually generated with 'make doc/core-en_US.xml'. ASTERISK-26639 Change-Id: Ifac562340c09f80c83e0203de098fcac93bf8c44
2017-10-25Merge "chan_sip: Fix SUBSCRIBE with missing "Expires" header." into 15Jenkins2
2017-10-25Merge "hashtab: Use ast_free." into 15Joshua Colp
2017-10-25Merge "res_pjsip_sdp_rtp: Fix setting of address type for rtp_ipv6" into 15Joshua Colp
2017-10-25Merge "res_pjproject.c: Upgrade bundled PJPROJECT to 2.7" into 15Joshua Colp
2017-10-24Merge "main/Makefile: Remove rule for non-existant testexpr2." into 15Jenkins2
2017-10-24Merge "res_xmpp: Ensure the connection filter is available." into 15Joshua Colp
2017-10-24Merge "lpc10: Avoid compiler warning when DONT_OPTIMIZE/COMPILE_DOUBLE." into 15Jenkins2
2017-10-24Merge "test_config: Fix failure and segfault when config_hook is run twice." ↵Joshua Colp
into 15
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 "chan_sip: Crypto attribute not last but first on SDP media level." ↵Joshua Colp
into 15
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
2017-10-23main/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
into 15
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'." into 15Jenkins2
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
2017-10-18res_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" into 15Jenkins2
2017-10-17Merge "cdr.c: Rename the Party A CDR container." into 15Joshua Colp
2017-10-17Merge "cdr.c: Add container to key off of Party B channel names." into 15Joshua Colp
2017-10-17Merge "contrib/script/sip_to_pjsip: implement 'all' for allow/disallow" into 15Joshua Colp
2017-10-17Merge "bridge_simple: Improve renegotiation success rate." into 15Jenkins2
2017-10-17Merge "chan_dahdi: wrap include file which is not present on BSD systems in ↵Joshua Colp
#ifdef" into 15
2017-10-17res_corosync: Fix linking issue with Corosync 2.xMatt Jordan
At some point in time in the history of Corosync (certainly within the 2.x branch), the corosync_cfg_state_track function was removed. Unfortunately, the cfg library is only linked if this function is present. Without the cfg library being linked to res_corosync, loading of res_corosync will fail. This patch makes it so that detecting corosync's core libraries, determined by the COROSYNC external library checks, links both the cpg and cfg libraries with res_corosync. Change-Id: I674e9e1c8fea11c3bf81154aaa7c1fd43f945465
2017-10-17Merge "res_pjsip_session: Rewrite o= with external_media_address." into 15Jenkins2
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
2017-10-16contrib/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
2017-10-16declare optional openssl dependencies in moduleinfoTzafrir Cohen
Declare optional openssl dependencies in: * res_rtp_asterisk.c * tcptls.c ASTERISK-27328 #close Change-Id: I2636f1c05b8104b4fe6f36cce0ebd9a98b9c78ab
2017-10-16Merge "cdr.c: Eliminated many calls to ao2_global_obj_ref()." into 15Joshua Colp
2017-10-15Merge "ast_bt_get_symbols: Prevent double-free." into 15Joshua Colp
2017-10-15Merge "tcptls: NULL-check the parameter of ast_ssl_teardown before accessing ↵Joshua Colp
it." into 15
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