asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	9d54dd04bb	Merge "res/res_pjsip: Add preferred_codec_only config to pjsip endpoint."	9 years ago
Aaron An	2a50c29101	res/res_pjsip: Add preferred_codec_only config to pjsip endpoint. This patch add config to pjsip by endpoint. ;preferred_codec_only=yes ; Respond to a SIP invite with the single most preferred codec ; rather than advertising all joint codec capabilities. This ; limits the other side's codec choice to exactly what we prefer. ASTERISK-26317 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Iad04dc55055403bbf5ec050997aee2dadc4f0762	9 years ago
zuul	51ec782372	Merge "res_pjsip_session: segfault on already disconnected session"	9 years ago
zuul	b5e4445b29	Merge "sorcery: Create function ast_sorcery_lockable_alloc."	9 years ago
zuul	825d6e036c	Merge "named_locks: Use ao2_weakproxy to deal with cleanup from container."	9 years ago
zuul	d57242a16b	Merge "astobj2: Support using a separate object for locking."	9 years ago
Alexei Gradinari	7bb7f7b9d5	res_pjsip_session: segfault on already disconnected session On heavy loaded system the TCP/TLS incoming calls could be disconnected by pjproject while these calls are being processed by asterisk which could use the session's memory pools. If the session in the disconnected state then the session memory pools were already freed, so we get segfault. This patch adds a lifetime control on an INVITE session to pjproject. The lifetime of the session is manipulated by calling pjsip_inv_add_ref/pjsip_inv_dec_ref. This patch uses these functions to inform pjproject that the session is in use. This patch adds check if the session state is not disconnected and also checks if the memory pool is not NULL. This patch also places tasks 'session_end' and 'session_end_completion' into session's serializer to avoid race condition. ASTERISK-26291 #close Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7	9 years ago
Joshua Colp	e34f299a96	Merge "codecs: Add Codec 2 mode 2400."	9 years ago
Corey Farrell	e875e1c12a	sorcery: Create function ast_sorcery_lockable_alloc. Create an alternative to ast_sorcery_generic_alloc which uses astobj2 shared locking. Use this new method for the 'struct ast_sip_aor' allocator. Change-Id: I3f62f2ada64b622571950278fbb6ad57395b5d6f	9 years ago
Corey Farrell	131baf70d6	named_locks: Use ao2_weakproxy to deal with cleanup from container. This allows standard ao2 functions to be used to release references to an ast_named_lock. This change can cause less frequent locking of the global named_locks container. The container is no longer locked when a named_lock reference is being release except when this causes the named_lock to be destroyed. Change-Id: I644e39c6d83a153d71b3fae77ec05599d725e7e6	9 years ago
Corey Farrell	0c5b6e9ff5	astobj2: Support using a separate object for locking. Create ao2_alloc_with_lockobj function to support shared locking. Change-Id: Iba687eb9843922be7e481e23a32c0700ecf88a80	9 years ago
Joshua Colp	4a8bdfc49b	Merge "res_fax: Fix deadlock in ast_channel_get_t38_state()."	10 years ago
Joshua Colp	179e8c15c8	Merge "res_fax: Fix deadlock setting FAXMODE channel variable."	10 years ago
zuul	c82cef8441	Merge "Fix checks for allocation debugging."	10 years ago
zuul	e3e08e1131	Merge "Fix naming mismatch of allocator functions."	10 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	10 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	10 years ago
Alexander Traud	2e79f52d71	codecs: Add Codec 2 mode 2400. ASTERISK-26217 #close Change-Id: I1e45d8084683fab5f2b272bf35f4a149cea8b8d6	10 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	10 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	10 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`)	10 years ago
George Joseph	534063fd67	res_pjsip: Add contact_user to endpoint contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4	10 years ago
Alexei Gradinari	e85adbd947	core: Entity ID is not set or invalid The Exchanging Device and Mailbox States could not working if the Entity ID (EID) is not set manually and can't be obtained from ethernet interface. This patch replaces debug message to warning and addes missing description about option 'entityid' to asterisk.conf.sample. With this patch the asterisk also: (1) decline loading the modules which won't work without EID: res_corosync and res_pjsip_publish_asterisk. (2) warn if EID is empty on loading next modules: pbx_dundi, res_xmpp Starting with v197 systemd/udev will automatically assign "predictable" names for all local Ethernet interfaces. This patch also addes some new ethernet prefixes "eno" and "ens". ASTERISK-26164 #close Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6	10 years ago
zuul	8d84c8edff	Merge "res_resolver_unbound: Allow compilation with libunbound version < 1.5"	10 years ago
zuul	74fffe9df2	Merge "res_srtp: Move SDP SRTP code from the core to res_srtp."	10 years ago
Richard Mudgett	41aba83ff6	res_srtp: Move SDP SRTP code from the core to res_srtp. A patch made to the master branch (Now the 14 branch) inadvertently made libsrtp a required dependency in order to compile Asterisk. Rather than create dummy defines to substitute for the defines supplied by libsrtp when libsrtp is not available, most of the code in sdp_srtp.c is moved into res_srtp.c. This gets more code out of Asterisk's core that isn't used when SRTP is not available. This also makes another inadvertent required dependency on libsrtp by Asterisk's core unlikely. ASTERISK-26253 #close Reported by: Ben Merrills Change-Id: I0a46cde81501c0405399c2588633ae32706d1ee7	10 years ago
Alexei Gradinari	820879415f	pjsip: Fix deadlock with suspend taskprocessor on masquerade If both channels which should be masqueraded are in the same serializer: 1st channel will be locked waiting condition 'complete' 2nd channel will be locked waiting condition 'suspended' On heavy load system a chance that both channels will be in the same serializer 'pjsip/distibutor' is very high. To reproduce compile res_pjsip/pjsip_distributor.c with DISTRIBUTOR_POOL_SIZE=1 Steps to reproduce: 1. Party A calls Party B (bridged call 'AB') 2. Party B places Party A on hold 3. Party B calls Voicemail app (non-bridged call 'BV') 4. Party B attended transfers Party A to voicemail using REFER. 5. When asterisk masquerades calls 'AB' and 'BV', a deadlock is happened. This patch adds a suspension indicator to the taskprocessor. When a session suspends/unsuspends the serializer it sets the indicator to the appropriate state. The session checks the suspension indicator before suspend the serializer. ASTERISK-26145 #close Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b	10 years ago
George Joseph	8d42ff784d	res_resolver_unbound: Allow compilation with libunbound version < 1.5 libunbound at version 1.4.20 (which CentOS still uses) declared all of their string function parameters as as 'char '. 1.4.21 changed them all to 'const char '. Thankfully 1.4.21 also introduced the UNBOUND_VERSION_MAJOR define so configure now checks for that and sets HAVE_UNBOUND_CONST_PARAMS. res_resolver_unbound then checks that and casts away the 'const' if it's not set. Tested compile and testsuite on CentOS6 (1.4.20), Ubuntu14 (1.4.22) and Fedora24 (1.5.4). There are a few failing tests to be addressed though. ASTERISK-26283 #close Change-Id: Ib708b19b706c5d0ba7b7d5473e6df339d9ae4148	10 years ago
zuul	393d571e93	Merge "Produce friendly error when AST_MODULE_SELF_SYM is not defined."	10 years ago
Corey Farrell	827457dca0	Produce friendly error when AST_MODULE_SELF_SYM is not defined. Modules must define AST_MODULE_SELF_SYM to be used as the name of a generated function. This produces a friendly error when it's not defined. ASTERISK-26278 #close Change-Id: Ib9d35a08104529c516d636771365e02c6e77a45b	10 years ago
Alexei Gradinari	403b63571c	res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack The PJSIP taskprocessors could be overflowed on startup if there are many (thousands) realtime endpoints configured with unsolicited mwi. The PJSIP stack could be totally unresponsive for a few minutes after boot completed. This patch creates a separate PJSIP serializers pool for mwi and makes unsolicited mwi use serializers from this pool. This patch also adds 2 new global options to tune taskprocessor alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. This patch also adds new global option 'mwi_disable_initial_unsolicited' to disable sending unsolicited mwi to all endpoints on startup. If disabled then unsolicited mwi will start processing on next endpoint's contact update. ASTERISK-26230 #close Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a	10 years ago
Corey Farrell	29b0f733a0	Add missing checks during startup. This ensures startup is canceled due to allocation failures from the following initializations. * channel.c: ast_channels_init * config_options.c: aco_init ASTERISK-26265 #close Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611	10 years ago
Richard Mudgett	68ebf86e2f	pbx.c: Allow dangerous functions when adding a hint to dialplan. We can allow dangerous functions when adding a hint since altering dialplan is itself a privileged activity. Otherwise, we could never execute dangerous functions. ASTERISK-25996 #close Reported by: Andrew Nagy Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba	10 years ago
zuul	e2bfcb3e58	Merge "codecs: Add iLBC 20."	10 years ago
George Joseph	8852a4c3db	asterisk.c: Add auto generation and persistence of UUID Upcoming features will require the generation and persistence of a UUID. Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d	10 years ago
Alexander Traud	8fb807009f	codecs: Add iLBC 20. Asterisk already supported iLBC 30. This change adds iLBC 20. Now, Asterisk defaults to iLBC 20 but falls back to iLBC 30, when the remote party requests this. ASTERISK-26218 #close ASTERISK-26221 #close Reported by: Aaron Meriwether Change-Id: I07f523a3aa1338bb5217a1bf69c1eeb92adedffa	10 years ago
Richard Mudgett	4286a369a1	res_pjsip: Whitespace and comment cleanup. Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38	10 years ago
zuul	9473818659	Merge "res_srtp: Enable AES-256 and AES-GCM."	10 years ago
Joshua Colp	7f36b79f87	Merge "res_fax: Fix FAXOPT(faxdetect) timeout option."	10 years ago
Joshua Colp	0933f0cf96	Merge "res_pjsip: Add fax_detect_timeout endpoint option."	10 years ago
zuul	194d0f606b	Merge "pbx: Create pbx_sw.c for management of 'struct ast_sw'."	10 years ago
zuul	fbdeb01edf	Merge "Add conditional support for noreturn functions."	10 years ago
Corey Farrell	a36a174c4b	pbx: Create pbx_sw.c for management of 'struct ast_sw'. This changes context switches from a linked list to a vector, makes 'struct ast_sw' opaque to pbx.c. Although ast_walk_context_switches is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_switches_count (AST_VECTOR_SIZE) * ast_context_switches_get (AST_VECTOR_GET) As with ast_walk_context_switches callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the switches, they have been converted to use the new functions. Change-Id: I08deb016df22eee8288eb03de62593e45a1f0998	10 years ago
Alexander Traud	1d2173c7ae	res_srtp: Enable AES-256 and AES-GCM. ASTERISK-26190 #close Change-Id: I11326d80edd656524a51a19450e586c583aa0a0b	10 years ago
Corey Farrell	8f6e9ffcc6	Add conditional support for noreturn functions. This adds support for tagging functions with the noreturn attribute. If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE and DO_CRASH are enabled then failed assertions never return. This can resolve a large number of false positives with static analyzers. ASTERISK-26220 #close Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753	10 years ago
Richard Mudgett	804fbd9c2b	res_fax: Fix FAXOPT(faxdetect) timeout option. The fax detection timeout option did not work because basically the wrong variable was checked in fax_detect_framehook(). As a result, the timer would timeout immediately and disable fax detection. * Fixed ignoring negative timeout values. We'd complain and then go right on using the negative value. * Fixed destroy_faxdetect() in the off-nominal case of an incomplete object creation. * Added more range checking to FAXOPT(gateway) timeout parameter. ASTERISK-26214 #close Reported by: Richard Mudgett Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976	10 years ago
Richard Mudgett	e739888d99	res_pjsip: Add fax_detect_timeout endpoint option. The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d	10 years ago
Corey Farrell	e2e8713b84	pbx: Create pbx_ignorepat.c for management of 'struct ast_ignorepat'. This changes context ignore patterns from a linked list to a vector, makes 'struct ast_ignorepat' opaque to pbx.c. Although ast_walk_context_ignorepats is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_ignorepats_count (AST_VECTOR_SIZE) * ast_context_ignorepats_get (AST_VECTOR_GET) As with ast_walk_context_ignorepats callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the ignorepats, they have been converted to use the new functions. Change-Id: I78f2157d275ef1b7d624b4ff7d770d38e5d7f20a	10 years ago
Corey Farrell	be36bd7ca5	pbx: Create pbx_include.c for management of 'struct ast_include'. This changes context includes from a linked list to a vector, makes 'struct ast_include' opaque to pbx.c. Although ast_walk_context_includes is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_includes_count (AST_VECTOR_SIZE) * ast_context_includes_get (AST_VECTOR_GET) As with ast_walk_context_includes callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the includes, they have been converted to use the new functions. const have been applied where possible to parameters for ast_include functions. Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60	10 years ago
Mark Michelson	273052f404	Update support for SILK format. This commit adds scaffolding in order to support the SILK audio format on calls. Roughly, this is what is added: * Cached silk formats. One for each possible sample rate. * ast_codec structures for each possible sample rate. * RTP payload mappings for "SILK". In addition, this change overhauls the res_format_attr_silk file in the following ways: * The "samplerate" attribute is scrapped. That's native to the format. * There are far more checks to ensure that attributes have been allocated before attempting to reference them. * We do not SDP fmtp lines for attributes set to 0. These changes make way to be able to install a codec_silk module and have it actually work. It also should allow for passthrough silk calls in Asterisk. Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e	10 years ago

1 2 3 4 5 ...

3513 Commits (c6aaf56be65ed9c0473b11c84a295c9361c01c14)