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Author	SHA1	Message	Date
Richard Mudgett	b86771d1bf	ast_framehook_detach() must be called with the channel locked. ... The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584	9 years ago
Richard Mudgett	5744f434f0	ast_framehook_attach() must be called with the channel locked. ... The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438	9 years ago
chrisderock	93b7533d74	app_macro: Consider '~~s~~' as a macro start extension. ... As described in issue ASTERISK-26282 the AEL parser creates macros with extension '~~s~~'. app_macro searches only for extension 's' so the created extension cannot be found. with this patch app_macro searches for both extensions and performs the right extension. ASTERISK-26282 #close Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb	9 years ago
varnav	d2e03c252d	chan_iax2: Set plaintext auth to deprecated as per ASTERISK-22820 ... Starting from draft 2 of RFC 5456 (October 23, 2006) plaintext auth is not supported in IAX2 protocol. Please refer to section 8.6.13 of RFC 5456. But plaintext auth is still supported by Asterisk implementation of IAX2. This support should be dropped. Patch, based on asterisk-dev discussion, adds deprecation warning on startup if 'auth' is set to 'plaintext', changes default values of 'auth' from 'md5, plaintext' to 'md5'. Patch is safe in terms of backwards compatibility, will work even if remote peers have auth=plaintext and we have defaults. auth=plaintext setting will remain deprecated in Asterisk 14 and 15, and IAX2 plaintext support will be removed in Asterisk 16. ASTERISK-22820 #close Change-Id: I5d2f3830cb57645604818f87518916e8a5c317bf	9 years ago
George Joseph	e40aa40aca	res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options ... ast_multicast_rtp_create_options now checks for NULL or empty options Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362	9 years ago
Alexander Traud	2e79f52d71	codecs: Add Codec 2 mode 2400. ... ASTERISK-26217 #close Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6	9 years ago
Mark Michelson	ded22c712a	ConfBridge: Rework announcer channel methodology ... NOTE: This patch was submitted earlier and reverted because of a failing test. The test has been patched so that it adjusts for the changes here, so this is being resubmitted for review. One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0	9 years ago
Joshua Colp	11ef7f34bf	Merge "Revert "ConfBridge: Rework announcer channel methodology""	9 years ago
Joshua Colp	065d810d3f	Revert "ConfBridge: Rework announcer channel methodology" ... This reverts commit 5aa8773052. Change-Id: I9ab45776e54a54ecf1bac9ae62d976dec30ef491	9 years ago
zuul	c9df806f24	Merge "ConfBridge: Rework announcer channel methodology"	9 years ago
zuul	4913fe3825	Merge "followme: initialize all config items on reload"	9 years ago
zuul	27813c7439	Merge "compilation failed with -Werror=maybe-uninitialized"	9 years ago
zuul	47c9acb5b2	Merge "res_odbc_transaction: add dep on generic_odbc"	9 years ago
Joshua Colp	9dd3d416cc	Merge "pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations."	9 years ago
Alexei Gradinari	41ee14bfae	compilation failed with -Werror=maybe-uninitialized ... The compilation failed for devmode --enable DONT_OPTIMIZE --enable BETTER_BACKTRACES --enable DO_CRASH --enable TEST_FRAMEWORK res_pjsip/pjsip_configuration.c: In function dtls_handler: res_pjsip/pjsip_configuration.c:974:20: error: back may be used uninitialized in this function [-Werror=maybe-uninitialized] int size = strlen(front); ^ cc1: all warnings being treated as errors Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580	9 years ago
zuul	d6b5f1b951	Merge "res_ari: Add http prefix to generated docs"	9 years ago
David M. Lee	eb0c9c476f	res_odbc_transaction: add dep on generic_odbc ... When res_odbc_transaction depended on res_odbc, it got the generic_odbc headers and libs implicitly. Now that it no longer depends on res_odbc, its dependency on generic_odbc must be explicit. Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911	9 years ago
Alexander Traud	12752c64cc	pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations. ... PJProject supports a lot of platforms even Windows, some with different defaults when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS, "/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows. Because of this, if configured with just an IPv6 address/transport, PJProject listens to both IPv4 and IPv6. However, this is not supported by the PJProject team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP, incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only server which accepts incoming connections on IPv4. If you try to configure two transports, one with IPv4 and one with IPv6 on the same interface, as expected by the PJProject team, the IPv4 transport is not able to bind because the IPv6 transport listens to both already. One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide. Then, you are able to configure two transports, one for each IP version on the same interface. That way, you get a server which works with IPv4 clients and IPv6 clients at the same time over the same interface. Here, this change sets this parameter directly within PJProject to match the expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack servers out of the box like in chan_sip. This change was accepted by the PJProject team as <http://trac.pjsip.org/repos/changeset/5403> and is expected to arrive in the next version, PJProject 2.6.0. Until then, this change is incorporated in the bundled PJProject of Asterisk. ASTERISK-26309 Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae	9 years ago
Corey Farrell	55ccdf93c3	Fix checks for allocation debugging. ... MALLOC_DEBUG should not be used to check if debugging is actually enabled, __AST_DEBUG_MALLOC should be used instead. MALLOC_DEBUG only indicates that debugging is requested, __AST_DEBUG_MALLOC indicates it is active. Change-Id: I3ce9cdb6ec91b74ee1302941328462231be1ea53	9 years ago
Corey Farrell	8061d9f66f	Fix naming mismatch of allocator functions. ... Allocator functions that take file/line/func parameters are prefixed with single-underscore when MALLOC_DEBUG is not defined, double-underscore when it is defined. This change updates all allocators that accept file/line/func to have the same prototype in either ABI mode. The parameter order of __ast_vasprintf and __ast_asprintf in utils.h have been changed to match that of astmm.h. End-use allocator macro's have been removed from astmm.h and moved to an unconditional part of utils.h. Change-Id: I823bb6ce2b5675b3a4735948f10a3b420e9a023a	9 years ago
zuul	c6ed91a9c8	Merge "sip_to_pjsip: Map externhost/ip to Transports."	9 years ago
Torrey Searle	c1b6a79686	res_ari: Add http prefix to generated docs ... updated the uri handler to include the url prefix of the http server this enables res_ari to add it to the uris when generating docs Change-Id: I279335a2625261a8492206c37219698f42591c2e (cherry picked from commit 6f448f32fe)	9 years ago
zuul	2b057d6215	Merge "res_odbc: Correct the dependency relationship with res_odbc_transaction"	9 years ago
zuul	1c64616373	Merge "sip.conf: tlsclientmethod is using sslv23 as default."	9 years ago
zuul	8932877044	Merge "rest-api: Swagger scripts were not replacing format variable in file brief"	9 years ago
zuul	be26a93687	Merge "sip_to_pjsip: Add cert_file."	9 years ago
zuul	22daced976	Merge "res_format_attr_g729: Add annexb=no format parameter to SDPs"	9 years ago
zuul	d86ee51ca0	Merge "res_pjsip: Add contact_user to endpoint"	9 years ago
zuul	50e0dfabc5	Merge "ari: Add documentation that path parameters are case-sensitive"	9 years ago
Alexander Traud	02a82f758e	sip_to_pjsip: Add cert_file. ... When using the migration script sip_to_pjsip.py, cert_file was not migrated to pjsip.conf. A previous change regarding this contained a copy/paste error. ASTERISK-22374 Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b	9 years ago
Alexander Traud	1a9555f036	sip.conf: tlsclientmethod is using sslv23 as default. ... When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL SSLv23_method. This was documented incorrectly in the file sip.conf.sample. SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that function should have been called 'secure_method' or 'automatic_method' back in the 90s. Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if you face a server which has problems like not falling back to TLSv1.0 automatically. ASTERISK-24425 Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3	9 years ago
Joshua Colp	b544bfbbd5	Merge "sip_to_pjsip: Write cos and tos."	9 years ago
Kevin Harwell	53a2f7dc88	res_format_attr_g729: Add annexb=no format parameter to SDPs ... Historically, Asterisk has always specified annexb=no for the g729 format. However, when using res_pjsip no format attribute was specified. This patch makes it so the SDP now contains a format attribute line with annexb=no. Note, that this means only g729a is negotiated. Even for pass through support. According to rfc7261 the type of annex used (a or b) is dependent upon the answerer. However, Asterisk being a back to back user agent makes this tricky to support at this time, thus we only allow annex 'a' for now. ASTERISK-26228 #close patches: res_format_attr_g729.c submitted by Jason Parker (license 4993) Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0	9 years ago
Kevin Harwell	7ea133f2ab	rest-api: Swagger scripts were not replacing format variable in file brief ... Given resource paths did not have 'json' substituted in for the '{format}'. For some auto generated documentation/comment strings it resulted in something like the following: "... REST handler for /api-docs/sounds.{format}" This patch makes sure the resource api's path is properly substituted. ASTERISK-25472 #close Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23	9 years ago
George Joseph	c7ffd6111d	res_odbc: Correct the dependency relationship with res_odbc_transaction ... The MODULEINFO dependencies between these 2 modules was reversed. res_odbc should depend on res_odbc_transaction, not the other way around. ASTERISK-25984 #close Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f	9 years ago
Kevin Harwell	966527249e	sip_to_pjsip: Set correct tls transport method ... A recent update had a copy/paste error where the unused variable 'val' was being passed to the set_value function instead of the 'method' value itself. This patch passes in the right variable. ASTERISK-22374 Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06	9 years ago
zuul	57f3e992e4	Merge "res_pjsip_session.c: Fix unbound srv failover tests."	9 years ago
Joshua Colp	2dba6d0371	Merge "sip_to_pjsip: Parse register even with transport."	9 years ago
Joshua Colp	71b3751093	Merge "sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit."	9 years ago
Joshua Colp	54c5bb0287	Merge "sip_to_pjsip: Map (session-)timers correctly."	9 years ago
Joshua Colp	5899b4c593	Merge "sip_to_pjsip: Add cert_file and ca_list_path."	9 years ago
Joshua Colp	560c2abdec	Merge "sip_to_pjsip: Write username even without authname."	9 years ago
Joshua Colp	14284aee45	Merge "sip_to_pjsip: Map the TLS method correctly."	9 years ago
Joshua Colp	0a09ab5b1c	Merge "sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent."	9 years ago
Joshua Colp	91624f439c	Merge "sip_to_pjsip: Write media_encryption."	9 years ago
Joshua Colp	b90ee04a99	Merge "sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry."	9 years ago
Mark Michelson	5aa8773052	ConfBridge: Rework announcer channel methodology ... One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5	9 years ago
Alexander Traud	e55d1e47aa	sip_to_pjsip: Map the TLS method correctly. ... When using the migration script sip_to_pjsip.py and tlsclientmethod is not set in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is offering/using not just TLSv1.0 but TLSv1.2 as well. ASTERISK-22374 Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f	9 years ago
Alexander Traud	da14c439a3	sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent. ... When using the migration script sip_to_pjsip.py, no section of type=system or type=general were created. Therefore the keys compactheaders, timerb, timert1, and useragent were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1	9 years ago
Alexander Traud	675721a7ab	sip_to_pjsip: Map (session-)timers correctly. ... When using the migration script sip_to_pjsip.py, session-timers=accept and session-timers=refuse were mapped to wrong values. ASTERISK-22374 Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092	9 years ago