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Author	SHA1	Message	Date
Matt Jordan	2d277996b7	Merge "Clang: Fix some more tautological-compare warnings." into 13	10 years ago
Matt Jordan	145f65598c	Merge "app_confbridge: Default the template option to a compatible default profile." into 13	10 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	10 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	10 years ago
Matt Jordan	bf3d9db4a6	Merge "res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX." into 13	10 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	10 years ago
Mark Michelson	b3cd5bc77f	Merge "Clang: change previous tautological-compare fixes." into 13	10 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	10 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	10 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	10 years ago
Matt Jordan	a06924e9d9	Merge "Astobj2: Ensure all calls to __adjust_lock pass a valid object." into 13	10 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	10 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	10 years ago
George Joseph	b0e929219b	.gitignore: Add .gcno and .gcda ... Products of --enable-coverage Change-Id: Ie20882d64b60692e2c941ea8872ab82a86ce77a3	10 years ago
Matt Jordan	5a3948a66f	Merge "Fix/Update clang-RAII macro implementation" into 13	10 years ago
Mark Michelson	2ef1e1fc68	Merge "res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers." into 13	10 years ago
Diederik de Groot	d6dfc85666	Clang: Fix some more tautological-compare warnings. ... 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: Ief23ef68916192b9b72dabe702b543ecfeca0b62	10 years ago
Joshua Colp	edd9e54818	Merge "Check for ao2_alloc failure in __ast_channel_internal_alloc." into 13	10 years ago
Joshua Colp	7b57116833	res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers. ... Currently the res_pjsip_mwi module only sends an unsolicited MWI NOTIFY upon a mailbox state change (such as a new message being left, or one being deleted). In practice this is not sufficient to keep clients aware of the current MWI status. This change makes the module send unsolicited MWI NOTIFY on startup so that clients are guaranteed to have the most up to date MWI information. It also makes clients receive an unsolicited MWI NOTIFY upon registration so if they are unaware of the current MWI status they receive it. ASTERISK-24982 #close Reported by: Joshua Colp Change-Id: I043f20230227e91218f18a82c7d5bb2aa62b1d58	10 years ago
Joshua Colp	4423d5f755	Merge "res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs." into 13	10 years ago
Corey Farrell	ad1a118632	Check for ao2_alloc failure in __ast_channel_internal_alloc. ... Fix a crash that could occur in __ast_channel_internal_alloc if ao2_alloc fails. ASTERISK-24991 #close Change-Id: I4ca89189eb22f907408cb87d0a1645cfe1314a90	10 years ago
Mark Michelson	3327560cb2	res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs. ... When SUBSCRIBE dialogs were established, we never associated the endpoint that created the subscription with the dialog we end up creating. In most cases, this ended up not causing any problems. The actual bug that was observed was that when a device that was behind NAT established a subscription with Asterisk, Asterisk would end up sending in-dialog NOTIFY requests to the device's private IP addres instead of the public address of the NAT router. When Asterisk receives the initial SUBSCRIBE from the device, res_pjsip_nat rewrites the contact to the public address on which the SUBSCRIBE was received. This allows for the dialog to have its target address set to the proper public address. Asterisk then would send a 200 OK response to the SUBSCRIBE, then a NOTIFY with the initial subscription state. The device would then send a 200 OK response to Asterisk's NOTIFY. Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat did not rewrite the address in the Contact header. Then, when the PJSIP dialog layer processed the 200 OK, PJSIP would perform a comparison between the IP address in the Contact header and its saved target address for the dialog. Since they differed, PJSIP would update the target dialog address to be the address in the Contact header. From this point, if Asterisk needed to send a NOTIFY to the device, the result was that the NOTIFY would be sent to the private address that the device placed in the Contact header. The reason why res_pjsip_nat did not rewrite the address when it received the 200 OK response was that it could not associate the incoming response with a configured endpoint. This is because on a response, the only way to associate the response to an endpoint is by finding the dialog that the response is associated with and then finding the endpoint that is associated with that dialog. We do not perform endpoint lookups on responses. res_pjsip_pubsub skipped the step of associating the endpoint with the dialog we created, so res_pjsip_nat could not find the associated endpoint and therefore couldn't rewrite the contact. This commit message is like 50x longer than the actual fix. ASTERISK 24981 #close Reported by Mark Michelson Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd	10 years ago
Richard Mudgett	d08446ec36	chan_dahdi/sig_pri: Make post AMI HangupRequest events on PRI channels. ... The chan_dahdi channel driver is a very old driver. The ability for it to support ISDN was added well after the initial analog support. Setting the softhangup flags is a carry over from the original analog code. The driver was not updated to call ast_queue_hangup() which will post the AMI HangupRequest event. * Changed sig_pri.c to call ast_queue_hangup() instead of setting the softhangup flag when the remote party initiates a hangup. ASTERISK-24895 #close Reported by: Andrew Zherdin Change-Id: I5fe2e48556507785fd8ab8e1c960683fd5d20325	10 years ago
Joshua Colp	96e18453f4	Merge "pjsip_options: Fix non-qualified contacts showing as unavailable" into 13	10 years ago
Diederik de Groot	2be9cc2643	Fix/Update clang-RAII macro implementation ... - When you need to refer to 'variable XXX' outside a block, it needs to be declared as '__block XXX', otherwise it will not be available with- in the block, making updating that variable hard to do, and ast_free lead to issues. - Removed the #error message because it creates complications when compiling external projects against asterisk For example when using a different compiler than the one used to compile asterisk. The warning/error should be generated during the configure process not the compilation process ASTERISK-24917 Change-Id: I12091228090e90831bf2b498293858f46ea7a8c2	10 years ago
George Joseph	b74b2cdcda	pjsip_options: Fix format specifier for int64_t rtt. ... Contact status rtt is an int64_t and needs the PRId64 macro to properly create the format specifier on 32-bit systems. Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7	10 years ago
Matt Jordan	27a122af66	Merge "main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple" into 13	10 years ago
Joshua Colp	9581a0ebf3	Merge "Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled." into 13	10 years ago
George Joseph	63169e00ff	pjsip_options: Fix non-qualified contacts showing as unavailable ... The "Add qualify_timeout processing and eventing" patch introduced an issue where contacts that had qualify_frequency set to 0 were showing Unavailable instead Unknown. This patch checks for qualify_frequency=0 and create an "Unknown" contact_status with an RTT = 0. Previously, the lack of contact_status implied Unknown but since we're now changing endpoint state based on contact_status, I've had to add new UNKNOWN status so that changes could trigger the appropriate contact_status observers. ASTERISK-24977: #close Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7	10 years ago
Matt Jordan	f0c82a173a	main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple ... When a PBX registrar is unloaded, it will fail to remove its extension from the context root_table if a dialplan application used by that extension is still loaded. This can be the case for AGI, which can be unloaded after several of the standard PBX providers. Often, this is harmless; however, if the extension's priorities are removed during the failed unloading *and* the dialplan application later unregisters, it leaves a ticking timebomb for the next PBX provider that attempts to iterate over the extensions. When that occurs, the peer_table pointer on the extension will already be set to NULL. The current code does not check to see if the pointer is NULL before passing it to a hashtab function this is not NULL tolerant. Since it is possible for the peer_table to be NULL when we normally would not expect that to be the case, the solution in this patch is to simply skip over processing an extension's priorities if peer_table is NULL. Prior to this patch, the tests/pbx/callerid_match test would crash during module unload. With this patch, the test no longer crashes after running. ASTERISK-24774 #close Reported by: Corey Farrell Change-Id: I2bbeecb7e0f77bac303a1b9135e4cdb4db6d4c40	10 years ago
Richard Mudgett	82bc0fd3ad	res_fax: Fix latent bug exposed by ASTERISK-24841 changes. ... Three fax related tests started failing as a result of changes made for ASTERISK-24841: tests/fax/pjsip/gateway_t38_g711 tests/fax/sip/gateway_mix1 tests/fax/sip/gateway_mix3 Historically, ast_channel_make_compatible() did nothing if the channels were already "compatible" even if they had a sub-optimal translation path already setup. With the changes from ASTERISK-24841 this is no longer true in order to allow the best translation paths to always be picked. In res_fax.c:fax_gateway_framehook() code manually setup the channels to go through slin and then called ast_channel_make_compatible(). With the previous version of ast_channel_make_compatible() this was always a no-operation. * Remove call to ast_channel_make_compatible() in fax_gateway_framehook() that now undoes what was just setup when the framehook is attached. * Fixed locking around saving the channel formats in fax_gateway_framehook() to ensure that the formats that are saved are consistent. * Fix copy pasta errors in fax_gateway_framehook() that confuses read and write when dealing with saved channel formats. ASTERISK-24841 Reported by: Matt Jordan Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d	10 years ago
Corey Farrell	c59a800707	Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled. ... When DEBUG_THREADLOCALS is enabled it causes the threadlocal cleanup to be called as a function. This causes a compile error with raw threadstorage as it uses NULL for cleanup. This fix uses a macro that provides NULL when DEBUG_THREADLOCALS is disabled, and replaces the call to "c_cleanup(data);" with "{};" when DEBUG_THREADLOCALS is enabled. ASTERISK-24975 #close Reported by: Ashley Sanders Change-Id: I3ef7428ee402816d9fcefa1b3b95830c00d5c402	10 years ago
Matt Jordan	e05b076827	Merge "Detect potential forwarding loops based on count." into 13	10 years ago
Mark Michelson	4f1a8dbe92	Detect potential forwarding loops based on count. ... A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23	10 years ago
George Joseph	674b18bdf0	pjsip_options: Add qualify_timeout processing and eventing ... This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Matt Jordan	f1abf51b73	Merge "res_pjsip: Refactor endpt_send_request to include transaction timeout" into 13	10 years ago
Matt Jordan	ab5b38e434	Merge "res_pjsip: Add global option to limit the maximum time for initial qualifies" into 13	10 years ago
Joshua Colp	ec77b6148f	Merge "res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced" into 13	10 years ago
Kevin Harwell	b56c1914fa	bridge.c: NULL app causes crash during attended transfer ... Due to a race condition there was a chance that during an attended transfer the channel's application would return NULL. This, of course, would cause a crash when attempting to access the memory. This patch retrieves the channel's app at an earlier time in processing in hopes that the app name is available. However, if it is not then "unknown" is used instead. Since some string value is now always present the crash can no longer occur. ASTERISK-24869 #close Reported by: viniciusfontes Review: Change-Id: I5134b84c4524906d8148817719d76ffb306488ac	10 years ago
Scott Griepentrog	8d4ce7cc2b	res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced ... This change makes the send_notify of the sub_tree not happen when the sub_tree has been deleted due to the notify call failing, which avoids a crash. ASTERISK-24970 #close Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf	10 years ago
George Joseph	bf46799f0e	res_pjsip: Refactor endpt_send_request to include transaction timeout ... This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html Since we currently have no control over pjproject transaction timeout, this patch pulls the pjsip_endpt_send_request function out of pjproject and into res_pjsip/endpt_send_transaction in order to implement that capability. Now when the transaction is initiated, we also schedule our own pj_timer with our own desired timeout. If the transaction completes before either timeout, pjproject cancels its timer, and calls our tsx callback where we cancel our timer and run the app callback. If the pjproject timer times out first, pjproject calls our tsx callback where we cancel our timer and run the app callback. If our timer times out first, we terminate the transaction which causes pjproject to cancel its timer and call our tsx callback where we run the app callback. Regardless of the scenario, pjproject is calling the tsx callback inside the group_lock and there are checks in the callback to make sure it doesn't run twice. As part of this patch ast_sip_send_out_of_dialog_request was created to replace its similarly named private function. It takes a new timeout argument in milliseconds (<= 0 to disable the timeout). ASTERISK-24863 #close Reported-by: George Joseph <george.joseph@fairview5.com> Tested-by: George Joseph <george.joseph@fairview5.com> Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747	10 years ago
George Joseph	1b6f6ff841	res_pjsip: Add global option to limit the maximum time for initial qualifies ... Currently when Asterisk starts initial qualifies of contacts are spread out randomly between 0 and qualify_timeout to prevent network and system overload. If a contact's qualify_frequency is 5 minutes however, that contact may be unavailable to accept calls for the entire 5 minutes after startup. So while staggering the initial qualifies is a good idea, basing the time on qualify_timeout could leave contacts unavailable for too long. This patch adds a new global parameter "max_initial_qualify_time" that sets the maximum time for the initial qualifies. This way you could make sure that all your contacts are initialy, randomly qualified within say 30 seconds but still have the contact's ongoing qualifies at a 5 minute interval. If max_initial_qualify_time is > 0, the formula is initial_interval = min(max_initial_interval, qualify_timeout * random(). If not set, qualify_timeout is used. The default is "0" (disabled). ASTERISK-24863 #close Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
George Joseph	5d218cde87	More .gitignore updates ... Added .pyc and .sha1 to the top-level .gitignore. Change-Id: I7dfc4f554d54d22947b38140d3305007503cc16a Tested-by: George Joseph <george.joseph@fairview5.com>	10 years ago
Matt Jordan	97f83c4c53	Merge "Build System: Replace comment about setting menuselect defaults." into 13	10 years ago
Rodrigo Ramírez Norambuena	abd56db3e0	cel_pgsql: Fix name string for log on unable allocate memory. ... The LOG_ERROR has reference to CDR instead of CEL for LENGTHEN_BUF1 and LENGTHEN_BUF2. ASTERISK-24965 #close Reported by: Rodrigo Ramirez Norambuena Change-Id: Icc818697d7d66d34bfe3048cdd15ca2b06c89744	10 years ago
Corey Farrell	222fbe1d9a	Build System: Replace comment about setting menuselect defaults. ... The Makefile claims that you can set default menuselect options by creating ~/.asterisk.makeopts or /etc/asterisk.makeopts, but those files have never been respected in Asterisk 11 or 13. This changes the comment to accurately reflect that these files are not automatically used by the build system. ASTERISK-13721 #close Reported by: pj Change-Id: Ibde804ff196283def49ccb9432fbf224a22586e2	10 years ago
Rodrigo Ramírez Norambuena	07e729cc7b	cdr_pgsql: Fix CLI "cdr show pgsql status" command. ... The command always showed the usage information. * Fix the error in command validation for CLI_SHOWUSAGE. ASTERISK-24959 #close Reported by: Rodrigo Ramirez Norambuena Change-Id: I584f0936bb01001336a468a55c1d05d79fe795d5 (cherry picked from commit 23a180cade)	10 years ago
George Joseph	7d43d85bea	.gitignore updates for master/13 ... Added products of ./bootstrap Added nmenuselect and gmenuselect to menuselect/ Change-Id: Ied658463958bafc04a9aff9ebc28e40c116a6e35	10 years ago
David M. Lee	3d27c223a5	Fixing extconf compile ... During the mass code deletion for clang support, a stray backslash was left behind that was causing utils to fail to compile. Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1	10 years ago
Matt Jordan	30045b4e67	Merge "build_tools/make_version: Update version parsing for Git migration" into 13	10 years ago