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Author	SHA1	Message	Date
Corey Farrell	63196a8256	res_pjsip_dlg_options: Fix MODULEINFO section. ... Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options. This extra space prevented any of the dependencies from being seen by menuselect, so building with default options would fail if PJSIP was not installed. This also makes the tool that extracts information for menuselect tolerant of multiple spaces in the future. ASTERISK-25033 #close Reported by: Peter Whisker Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698	11 years ago
Corey Farrell	ac1f0090eb	Build System: Prevent unneeded changes to asterisk/buildopts.h. ... * Add AST_DEVMODE to BUILDOPTS * Remove CFLAGS that do not effect ABI from BUILDOPTS. * Use BUILDOPTS to generate AST_BUILDOPT_SUM. * Remove loop that defined AST_MODULE_* These changes ensure that only ABI effecting options are considered for AST_BUILDOPT_SUM. This also reduces unneeded full system rebuilds caused by enabling or disabling one module that another is dependent on. ASTERISK-25028 Reported by: Corey Farrell Change-Id: I2c516d93df9f6aaa09ae079a8168c887a6ff93a2	11 years ago
Corey Farrell	5875bf183c	Astobj2: Fix initialization order of refdebug and AO2_DEBUG. ... This ensures that refdebug is initialized before AO2_DEBUG if both are enabled, since AO2_DEBUG allocates a container. This change also makes AO2_DEBUG initialization critical, a failure will abort Asterisk startup. This is needed since the failure would be caused by reg_containers allocation failure, and that would result in a segmentation fault by ao2_container_register later in startup. ASTERISK-25048 #close Reported by: Corey Farrell Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244	11 years ago
Matt Jordan	1b19c15f17	main/pbx: Improve performance of dialplan reloads with a large number of hints ... The PBX core maintains two hash tables for hints: a container of the actual hints (hints), along with a container of devices that are watching that hint (hintdevices). When a dialplan reload occurs, each hint in the hints container is destroyed; this requires a lookup in the container of devices to find the device => hint mapping object. In the current code, this performs an ao2_callback, iterating over each of the device to hint objects in the hintdevices container. For a large number of hints, this is extremely expensive: dialplan reloads with 20000 hints could take several minutes in just this phase. This patch improves the performance of this step in the dialplan reloads by caching which devices are watching a hint on the hint object itself. Since we don't want to create a circular reference, we just cache the name of the device. This allows us to perform a smarter ao2_callback on the hintdevices container during hint removal, hashing on the name of the device and returning an iterator to the matching names. The overall performance improvement is rather large, taking this step down to a number of seconds as opposed to minutes. In addition, this patch also registers the hint containers in the PBX core with the astobj2 library. This allows for reasonable debugging to hash collisions in those containers. ASTERISK-25040 #close Reported by: Matt Jordan Change-Id: Iedfc97a69d21070c50fca42275d7b3e714e59360	11 years ago
Matt Jordan	ec0f80b6e8	Merge "res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback." into 13	11 years ago
Matt Jordan	ed51fbbe9c	Merge "Prevent potential crash on blond transfer." into 13	11 years ago
Corey Farrell	3efe0df044	Sample Configs: Fix syntax error in pjsip.conf ... The sample pjsip.conf has a few comment lines that are missing the semicolons at the start of the comment, causing the config to fail load. Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352	11 years ago
Mark Michelson	077979618b	Prevent potential crash on blond transfer. ... Scenario: Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects the incoming call (or some other immediate circumstance causes Carol not to answer the call) What occurs in this case is that when the bridge between Alice and Bob breaks, Alice is told to masquerade into Bob's channel that had placed the call to Carol. The actual masquerade goes down without a hitch. However, a channel fixup callback that attempts to publish dial events over Stasis has a crash. The reason for this crash is that the datastore on Bob's channel that placed the outbound call to Carol only had a bare pointer to Carol's channel. Since Carol rejected the incoming call, Carol's channel has been hung up and freed, meaning accessing her channel results in a crash. The fix here is simple. The dial fixup code has been altered to hold references to the involved channels and to drop those references when freeing data. ASTERISK-25025 #close Reported by Chet Stevens Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197	11 years ago
Mark Michelson	4b8cddfb36	res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback. ... The Asterisk 13 version of the fix for outbound registration was missing a key component that set the outbound authenticator's callback that creates an authenticated request based on an old request. This was picked up by some outbound registration tests failing in the testsuite. Change-Id: I5ca9379698c606da36bc38eaffccedaf64211ce3	11 years ago
Joshua Colp	415a0d0745	res_ari_device_states: Fix dependency on res_stasis_device_state. ... The res_ari_device_states module depends on res_stasis_device_state, not res_stasis_device_states. Change-Id: I26e02ad37f9e36bcc859867e2fad1b90452ec3de	11 years ago
Mark Michelson	e0c6f88010	Merge "chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option." into 13	11 years ago
Matt Jordan	d1bc86fc99	Merge "res_pjsip_outbound_registration: Add virtual line support." into 13	11 years ago
Richard Mudgett	d3c310a28c	chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option. ... Some telco switches occasionally ignore ISDN RESTART requests. The fix for ASTERISK-19608 added an escape clause for B channels in the restarting state if the telco ignores a RESTART request. If the telco fails to acknowledge the RESTART then Asterisk will assume the telco acknowledged the RESTART on the second call attempt requesting the B channel by the telco. The escape clause is good for dealing with RESTART requests in general but it does cause the next call for the restarting B channel to be rejected if the telco insists the call must go on that B channel. chan_dahdi doesn't really need to issue a RESTART request in response to receiving a cause 44 (Requested channel not available) code. Sending the RESTART in such a situation is not required (nor prohibited) by the standards. I think chan_dahdi does this for historical reasons to deal with buggy peers to get channels unstuck in a similar fashion as the chan_dahdi.conf resetinterval option. * Add the chan_dahdi.conf force_restart_unavailable_chans compatability option that when disabled will prevent chan_dahdi from trying to RESTART the channel in response to a cause 44 code. ASTERISK-25034 #close Reported by: Richard Mudgett Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65	11 years ago
Rodrigo Ramírez Norambuena	7f611fa0e8	cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8 ... This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR columns added in Asterisk 1.8. The columns are: * peeraccount * linkedid * sequence When enabled, the columns in the database entry will be populated with the data from the CDR. ASTERISK-24976 #close Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b	11 years ago
Joshua Colp	e332c7ed5e	res_pjsip_outbound_registration: Fix double unref on error return. ... When the PJSIP pjsip_regc_send function is invoked and an error status returned the caller currently decrements the reference count of the client state that it just incremented, assuming the registration callback would not have been invoked. In practice this is not correct. If the failure happens after the transaction has been set up the callback will still be invoked. This will cause the reference count to be incorrectly decremented twice, once by the registration callback and second by the caller of pjsip_regc_send. This change makes it so that whether the callback is invoked or not is known by the caller of pjsip_regc_send. Depending on this it can know whether it is responsible for decrementing the reference count of the client state or not. ASTERISK-25037 #close Reported by: Joshua Colp Change-Id: I749dc12f3a22115c49c5d7d95ff42a5fa45319de	11 years ago
Diederik de Groot	9c3ed42875	Update configure.ac/Makefile for clang ... Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which checks compiler requirements for RAII: gcc: -fnested-functions support clang: -fblocks (and if required -lBlocksRuntime) The original check was implemented in configure.ac and now has it's own file. This function also sets C_COMPILER_FAMILY to either gcc or clang for use by makefile Created autoconf/ast_check_strsep_array_bounds.m4 (contains AST_CHECK_STRSEP_ARRAY_BOUNDS): which checks if clang is able to handle the optimized strsep & strcmp functions (linux). If not, the standard libc implementation should be used instead. Clang + the optimized macro's work with: strsep(char *, char []), but not with strsepo(char *, char *). Instead of replacing all the occurences throughout the source code, not using the optimized macro version seemed easier See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h': llvm-comment: Normally, this array-bounds warning are suppressed for macros, so that unused paths like the one that accesses __s1[3] are not warned about. But if you preprocess manually, and feed the result to another instance of clang, it will warn about all the possible forks of this particular if statement. Instead of switching of this optimization, another solution would be to run the preproces- sing step with -frewrite-includes, which should preserve enough information so that clang should still be able to suppress the diag- nostic at the compile step later on. See also "https://llvm.org/bugs/show_bug.cgi?id=20144" See also "https://llvm.org/bugs/show_bug.cgi?id=11536" Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning suppressions: -Wno-unused-value -Wno-parentheses-equality In an earlier review (reviewboard: 4550 and 4554), they were deemed a nuisace and less than benefitial. configure.ac: Added AST_CHECK_RAII() see earlier Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier Removed moved content ASTERISK-24917 Change-Id: I12ea29d3bda2254ad3908e279b7effbbac6a97cb	11 years ago
Matt Jordan	37a193da18	Merge "ARI: Fix missing dependencies." into 13	11 years ago
Matt Jordan	6a86b3555b	Merge "res_fax: allow 2400 transmission rate according to v.27ter standard" into 13	11 years ago
Matt Jordan	d4e207e27e	main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8 ... The patch in 0b6410c4f8 did correctly fix a memory leak of the DTLS structures in the RTP engine. However, when a 'core reload' is issued, a double free of the memory pointed to by the char *'s in the DTLS configuration struct can occur, as ast_rtp_dtls_cfg_free does not set the pointers to NULL when they are freed. This patch sets those pointers to NULL, preventing a second call to ast_rtp_dtls_cfg_free from corrupting memory. ASTERISK-25022 Change-Id: I820471e6070a37e3c26f760118c86770e12f6115	11 years ago
Kevin Harwell	3fb6daeb55	res_fax: allow 2400 transmission rate according to v.27ter standard ... A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits per second. This reverts all or some of those patches since according to the v.27ter standard a rate of 2400 bits per second is also supported. One of the original patches also added 9600 bits per second support for v.27. This patch also removes that since v.27ter only supports 2400/4800 bits per second. Also, since Asterisk specifically supports v.27ter the enum was renamed to better reflect this. ASTERISK-24955 #close Reported by: Matt Jordan Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733	11 years ago
Joshua Colp	49ef81c15c	res_sorcery_config: Fix build issue due to syntax error. ... Change-Id: Ic8322f04e37842848ad72cf2871bd0378f67c4ac	11 years ago
Ashley Sanders	3278fe5327	chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR ... Sections Exist in pjsip.conf This patch modifies the current loading strategy of the pjsip configuration. If duplicate sections (e.g. sections containing the same [id/type]) are defined in [pjsip.conf], the loader will consider the configuration for the given type as invalid when the duplicate section is encountered. The entire configuration (including what was previously loaded) for the duplicate [id/type] sections will be rejected and destroyed, an error message is logged and the load processing for the given stops. ASTERISK-24996 Reported By: Ashley Sanders Change-Id: I35090ca4cd40f1f34881dfe701a329145c347aef	11 years ago
Joshua Colp	89f6719f7a	res_pjsip_outbound_registration: Add virtual line support. ... Virtual line support establishes a relationship between messages related to an outbound registration and a local endpoint. This is accomplished by attaching a parameter to the Contact of the outbound registration and looking for it on any received requests. If the parameter exists and can be matched to an outbound registration the configured endpoint is associated with the request. ASTERISK-24949 #close Reported by: Joshua Colp Change-Id: I7df909d2625479110a83fdd354c21ac539e8615d	11 years ago
Corey Farrell	d61f03c4f9	ARI: Fix missing dependencies. ... ARI modules that are generated by 'make ari-stubs' are all dependent on res_ari_model. Additionally some of the same modules depend on one or more res_stasis_* modules. ASTERISK-25027 #close Reported by: Corey Farrell Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153	11 years ago
Corey Farrell	3e4624ad21	res_pjsip: Remove incorrect MODULEINFO from presence_xml.c. ... Remove incorrect MODULEINFO block and unneeded header includes from presence_xml.c. ASTERISK-25027 Reported by: Corey Farrell Change-Id: I977c609ab9d1fe05373027c4138900f6985990eb	11 years ago
Corey Farrell	fed9faab8d	Git Migration: Create doc/rest-api when needed. ... Create the directory './doc/rest-api' at the start of 'make ari-stubs' to prevent an error when documentation is generated. The directory is also added to git ignores. ASTERISK-25027 Reported by: Corey Farrell Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b	11 years ago
Joshua Colp	df23c8a86b	res_pjsip_outbound_registration: Fix build due to removal of transaction. ... Change-Id: I7a8a7beec3334cec304943f2dd7597eabe2e3150	11 years ago
Joshua Colp	95ab9fdb1a	Merge "res_pjsip_outbound_registration: Add debugging messages." into 13	11 years ago
Joshua Colp	0e70dc0dc8	Merge "res_pjsip_outbound_registration: Don't fail on delayed processing: 13." into 13	11 years ago
Mark Michelson	e39bd6ba46	res_pjsip_outbound_registration: Don't fail on delayed processing: 13. ... This is the Asterisk 13 version of a change to master that allows for registration responses to be processed successfully potentially after the original transaction has timed out. The main difference between this and the master change is that the master version has API changes that are unacceptable for 13. For 13, this is worked around by adding a new API call that the outbound registration code uses instead. The following is the text from the master version of this commit: Odd behaviors have been observed during outbound registrations. The most common problem witnessed has been one where a request with authentication credentials cannot be created after receiving a 401 response. Other behaviors include apparently processing an incorrect SIP response. Inspecting the code led to an apparent issue with regards to how we handle transactions in outbound registration code. When a response to a REGISTER arrives, we save a pointer to the transaction and then push a task onto the registration serializer. Between the time that we save the pointer and push the task, it's possible for the transaction to be destroyed due to a timeout. It's also possible for the address to be reused by the transaction layer for a new transaction. To allow for authentication of a REGISTER request to be authenticated after the transaction has timed out, we now also hold a reference to the original REGISTER request instead of the transaction. The function for creating a request with authentication has been altered to take the original request instead of the transaction where the original request was sent. ASTERISK-25020 Reported by Mark Michelson Change-Id: If1ee5f601be839479a219424f0358a229f358f7c	11 years ago
Mark Michelson	1bf008fc76	res_pjsip_outbound_registration: Add debugging messages. ... When problems occur regarding outbound registrations, it currently is difficult to debug. Most off-nominal paths had warning messages, but sometimes we want to know what's going on before hitting the off-nominal path. This patch adds lots of debugging output that should give a clearer picture of what is happening with regards to outbound registrations. ASTERISK-25020 Reported by Mark Michelson Change-Id: I577bde7860be0a6c872b5bcb4d5047340bf45d45	11 years ago
Joshua Colp	7ee05892d6	Merge "Example script for scan-build (the llvm static analyzer)" into 13	11 years ago
Steve Davies	0b6410c4f8	res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS ... ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created. The resources are linked into a table, but the original alloc refs are never released. ast_strdup leak in rtp_engine.c. If ast_rtp_dtls_cfg_copy() is called twice on the same destination struct, a pointer to an alloc'd string is overwritten before the string is free'd. ASTERISK-25022 Reported by: one47 Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b	11 years ago
Joshua Colp	427209603d	Merge "cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version" into 13	11 years ago
George Joseph	99fb87ae13	res_pjsip: Fix SEGV on pending-qualify contacts ... Permanent contacts that hadn't been qualified yet were missing their contact_status entries causing SEGVs when running CLI commands. This patch makes sure that contact_statuses are created for both dynamic and permanent contacts when they are created. It also adds checks in the CLI code to make sure there's a contact_status, just in case. ASTERISK-25018 #close Reported-by: Ivan Poddubny Tested-by: Ivan Poddubny Tested-by: George Joseph Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029	11 years ago
Rodrigo Ramírez Norambuena	d5dd43856e	cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version ... Add new column to INSERT new columns added in cdr 1.8 version. The columns are: * peeraccount * linkedid * sequence This feature is configurable in cdr_odbc.conf using a new configuration option, 'newcdrcolumns'. ASTERISK-24976 #close Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127	11 years ago
Matt Jordan	e9788056e9	channels/chan_skinny: Fix compilation error introduced in f8e21a1adf ... A typo in commit f8e21a1adf resulted in a compilation error in chan_skinny. This patch fixes the typo. ASTERISK-24917 Change-Id: Id7f4ad1fe948eb2408622e80c27936ce4516c33c	11 years ago
Matt Jordan	2d277996b7	Merge "Clang: Fix some more tautological-compare warnings." into 13	11 years ago
Matt Jordan	145f65598c	Merge "app_confbridge: Default the template option to a compatible default profile." into 13	11 years ago
Kevin Harwell	7e5056b393	app_confbridge: Default the template option to a compatible default profile. ... Confbridge dynamic profiles did not have a default profile unless you explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a template was not set prior to the bridge being created then some options were left with no default values set. This patch makes it so the default templates are set to the default bridge and user profiles. ASTERISK-24749 #close Reported by: philippebolduc Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a	11 years ago
Mark Michelson	1da9ec969d	res_pjsip_outbound_authenticator: Increase CSeq on authed requests. ... The way PJSIP generates an authenticated request is to use a previous request as a template. This means that the authenticated request will have the same Call-ID, From header (including tag), and CSeq as the original request. PJSIP generates a new branch on the Via header to indicate that this is a new transaction, though. There are some SIP implementations, though, that do not notice the change in the branch and therefore will match the authed request to the original request's transaction. Since the CSeq is the same, the server will repeat the response it sent to the original request. This patch aids interoperability by increasing the CSeq of the authed request by one. ASTERISK-24845 #close Reported by: Carl Fortin Tested by: Carl Fortin Change-Id: I39c4ca52e688a9f83bcc1878371334becdc5be01	11 years ago
Matt Jordan	bf3d9db4a6	Merge "res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX." into 13	11 years ago
Diederik de Groot	cb318f3960	Example script for scan-build (the llvm static analyzer) ... - Added Pre-amble (Options / Flags / Usage Example / GNU License) - Extended Configurability - Made Executable ASTERISK-24917 Change-Id: I70405fe54e4be7dbfbcb62e291690069b88617a8	11 years ago
Mark Michelson	b3cd5bc77f	Merge "Clang: change previous tautological-compare fixes." into 13	11 years ago
Mark Michelson	eabf3b5a3c	res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX. ... When Asterisk originates a channel to an application, the channel is hung up once the application finishes executing. When the application in question is SendFax, the Asterisk PJSIP code will attempt to reinvite the T.38 session to audio after the FAX completes. The hangup of the channel happens in the midst of this reinvite transaction. In most circumstances, this works out okay because the BYE is delayed until the reinvite transaction can complete. However, if the reinvite that Asterisk sends receives a 401/407 response, then Asterisk's attempt to re-send the reinvite with authentication will fail. This is because the session supplement in res_pjsip_t38 makes the assumption that the channel on the session will always be non-NULL. Since the channel has been hung up, though, the channel is now NULL. Attempting to operate on the channel causes a crash. This patch fixes the issue by ensuring that the channel on the session is not NULL before attempting to mess with the T.38 framehook. This patch also contains some corrections for comments that were incorrect and really confused me when I first started looking at the code. ASTERISK-25004 #close Reported by Mark Michelson Change-Id: Ic5a1230668369dda4bb13524098aed9306ab45a0	11 years ago
George Joseph	f70d21b2cf	res_pjsip: Validate that contact uris start with sip: or sips: ... Currently we use pjsip_parse_hdr to validate contact uris but it appears that it allows uris without a scheme if there's a port supplied. I.E myexample.com will fail but myexample.com:5060 will pass even though it has no scheme. This causes SEGVs later on whenever the uri is used. To prevent this, permanent_contact_validate has been updated to check that the scheme is either 'sip' or 'sips'. 2 uses of possibly-null endpoint have also been fixed in create_out_of_dialog_request. ASTERISK-24999 Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2 Reported-by: Brad Latus	11 years ago
Diederik de Groot	1bb16bedc7	Clang: change previous tautological-compare fixes. ... clang can warn about a so called tautological-compare, when it finds comparisons which are logically always true, and are therefor deemed unnecessary. Exanple: unsigned int x = 4; if (x > 0) // x is always going to be bigger than 0 Enum Case: Each enumeration is its own type. Enums are an integer type but they do not have to be *signed*. C leaves it up to the compiler as an implementation option what to consider the integer type of a particu- lar enumeration is. Gcc treats an enum without negative values as an int while clang treats this enum as an unsigned int. rmudgett & mmichelson: cast the enum to (unsigned int) in assert. The cast does have an effect. For gcc, which seems to treat all enums as int, the cast to unsigned int will eliminate the possibility of negative values being allowed. For clang, which seems to treat enums without any negative members as unsigned int, the cast will have no effect. If for some reason in the future a negative value is ever added to the enum the assert will still catch the negative value. ASTERISK-24917 Change-Id: I0557ae0154a0b7de68883848a609309cdf0aee6a	11 years ago
Matt Jordan	a06924e9d9	Merge "Astobj2: Ensure all calls to __adjust_lock pass a valid object." into 13	11 years ago
George Joseph	1474bb05f6	res_corosync: Add check for config file before calling corosync apis ... On some systems, res_corosync isn't compatible with the installed version of corosync so corosync_cfg_initialize fails, load_module returns LOAD_FAILURE, and Asterisk terminates. The work around has been to remember to add res_corosync as a noload in modules.conf. A better solution though is to have res_corosync check for its config file before attempting to call corosync apis and return LOAD_DECLINE if there's no config file. This lets Asterisk loading continue. If you have a res_corosync.conf file and res_corosync fails, you get the same behavior as today and the fatal error tells you something is wrong with the install. ASTERISK-24998 Change-Id: Iaf94a9431a4922ec4ec994003f02135acfdd3889	11 years ago
Corey Farrell	73efb093b8	Astobj2: Ensure all calls to __adjust_lock pass a valid object. ... __adjust_lock doesn't check for invalid objects, and doesn't have an appropriate return value for invalid objects. Most callers of __adjust_lock pass objects that have already been confirmed valid, this change adds checks before the remaining calls. ASTERISK-24997 #close Reported by: Corey Farrell Change-Id: I669100f87937cc3f867cec56a27ae9c01292908f	11 years ago