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Author	SHA1	Message	Date
zuul	795532b2d5	Merge "ast_framehook_attach() must be called with the channel locked."	9 years ago
zuul	c82cef8441	Merge "Fix checks for allocation debugging."	9 years ago
zuul	e3e08e1131	Merge "Fix naming mismatch of allocator functions."	9 years ago
Alexander Traud	858fa5eb2c	channel: No hung-up on failing security requirements. ... In your Diaplan, if you specify same => n,Set(CHANNEL(secure_bridge_media)=1) same => n,Set(CHANNEL(secure_bridge_signaling)=1) only the SIP channel driver chan_sip supports this. All other channels drivers like res_pjsip fail. In case of failure, the original sRTP source code released the whole channel, even if not hung-up, yet. This change does not release the channel but instead hangs-up the channel. ASTERISK-26306 Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db	9 years ago
Alexander Traud	f35501b8c9	sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations. ... When using the migration script sip_to_pjsip.py, and your sip.conf is configured with bindaddr=::, two transports are written to pjsip.conf, one for 0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4 and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface like in chan_sip. Furthermore, the script internal functions "build_host" and "split_hostport" did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change makes sure, even such addresses are parsed correctly. ASTERISK-26309 Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48	9 years ago
Richard Mudgett	ea929d766d	res_pjsip: Cache global config options. ... We may check a global config option hundreds of times a second or more. Asking sorcery for the global configuration from the config files backend involves several allocations and container traversals. Using realtime without a memory cache is a lot worse because you have to lookup in the realtime database each time to reconstitute the sorcery object. With a memory cache for realtime, there is about the same amount of overhead as for config files. Either way, it is still fairly expensive to access the sorcery object that much. * Cache the global config options so we can access them faster. You must now always perform a res_pjsip reload to change the global options. Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7	9 years ago
Richard Mudgett	5eb6cb969f	res_fax: Fix deadlock in ast_channel_get_t38_state(). ... ast_channel_get_t38_state() calls ast_channel_queryoption() with AST_OPTION_T38_STATE. If the passed in channel is a local channel then a deadlock can happen if a channel lock is held when called. * Made ast_channel_get_t38_state() callers not hold a channel lock before calling. * Update ast_channel_get_t38_state() doxygen to note that no channel locks can be held when calling the function. ASTERISK-26203 #close Reported by: Etienne Lessard ASTERISK-24822 #close Reported by: David Brillert ASTERISK-22732 #close Reported by: Richard Mudgett Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214	9 years ago
Richard Mudgett	277a2d667a	res_fax: Fix deadlock setting FAXMODE channel variable. ... ASTERISK-25980 added the FAXMODE channel variable to res_fax.c. Unfortunately, it also introduced a deadlock potential because set_channel_variables() which sets FAXMODE can be called during a masquerade. The ast_channel_get_t38_state() which gets the value used to set FAXMODE cannot be called with the channel locked. As a result, local channels can deadlock because of how they must acquire the locks necessary to operate. The intent of FAXMODE is for dialplan to know how a fax was transferred after the fax completes. However, the previous patch sets FAXMODE to the channel's current T.38 state AFTER the fax has completed and where T.38 may have already disconnected. * Set FAXMODE based upon T.38 negotiations exchanged either with the fax applications or the fax framehooks. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1	9 years ago
Richard Mudgett	edca14c8a5	res_fax.c: Fix deadlock in fax_gateway_indicate_t38(). ... fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be called with any channel locks already held. A deadlock can happen if the function is operating on a local channel. * Made fax_gateway_indicate_t38() unlock the channel before calling ast_indicate_data() since fax_gateway_indicate_t38() is always called with the channel locked. * Made fax_gateway_indicate_t38() return void since nothing cared about its return value. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407	9 years ago
Richard Mudgett	141cd42880	res_fax.c: Add chan locked precondition comments. ... Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7	9 years ago
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