26702 Commits (1ad9a6b6b60e841b3cadda31fc786ac42f74133f)

 

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Author	SHA1	Message	Date
Alexander Traud	1ad9a6b6b6	chan_sip: Reload peer without its old capabilities. ... On reload, previously allowed codecs were not removed. Therefore, it was not possible to remove codecs while Asterisk was running. Furthermore, newly added codecs got appended behind the previous codecs. Therefore, it was not possible to add a codec with a priority of #1. This change removes the old capabilities before the current ones are added. ASTERISK-25182 #close Reported by: Alexander Traud patches: asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520) Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802	10 years ago
Joshua Colp	5caefc98a1	chan_sip: Destroy peers without holding peers container lock. ... Due to the use of stasis_unsubscribe_and_join in the peer destructor it is possible for a deadlock to occur when an event callback is occurring at the same time. This happens because the peer may be destroyed while holding the peers container lock. If this occurs the event callback will never be able to acquire the container lock and the unsubscribe will never complete. This change makes it so the peers that have been removed from the peers container are not destroyed with the container lock held. ASTERISK-25163 #close Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33	10 years ago
Matt Jordan	bd77ace25a	Merge "Resolve race conditions involving Stasis bridges."	10 years ago
Mark Michelson	d7a1e84a1e	Resolve race conditions involving Stasis bridges. ... This resolves two observed race conditions. First, a bit of background on what the Stasis application does: 1a Creates a stasis_app_control structure. This structure is linked into a global container and can be looked up using a channel's unique ID. 2a Puts the channel in an event loop. The event loop can exit either because the stasis_app_control structure has been marked done, or because of some other factor, such as a hangup. In the event loop, the stasis_app_control determines if any specific ARI commands need to be run on the channel and will run them from this thread. 3a Checks if the channel is bridged. If the channel is bridged, then ast_bridge_depart() is called since channels that are added to Stasis bridges are always imparted as departable. 4a Unlink the stasis_app_control from the container. When an ARI command is received by Asterisk, the following occurs 1b A thread is spawned to handle the HTTP request 2b The stasis_app_control(s) that corresponds to the channel(s) in the request is/are retrieved. If the stasis_app_control cannot be retrieved, then it is assumed that the channel in question has exited the Stasis app or perhaps was never in Stasis in the first place. 3b A command is queued onto the stasis_app_control, and the channel's event loop thread is signaled to run the command. 4b While most ARI commands do nothing further, some, such as adding or removing channels from a bridge, will block until the command they issued has been completed by the channel's event loop. The first race condition that is solved by this patch involves a crash that can occur due to faulty detection of the channel's bridged status in step 3a. What can happen is that in step 2a, the event loop may run the ast_bridge_impart() function to asynchronously place the channel into a bridge, then immediately exit the event loop because the channel has hung up. In step 3a, we would detect that the channel was not bridged and would not call ast_bridge_depart(). The reason that the channel did not appear to be bridged was that the depart_thread that is spawned by ast_bridge_impart() had not yet started. That is the thread where the channel is marked as being bridged. Since we did not call ast_bridge_depart(), the Stasis application would exit, and then the channel would be destroyed Then the depart_thread would start up and try to manipulate the destroyed channel, causing a crash. The fix for this is to switch from using ast_channel_is_bridged() to checking the NULLity of ast_channel_internal_bridge_channel() to determine if ast_bridge_depart() needs to be called. The channel's internal bridge_channel is set when ast_bridge_impart() is called and is NULLed by the call to ast_bridge_depart(). If the channel's internal bridge_channel is non-NULL, then the channel must have been imparted into the bridge and needs to be departed, even if the actual bridging operation has not yet started. By departing the channel when necessary, the thread that is running the Stasis application will block until the bridge gives the okay that the depart_thread has exited. The second race condition that is solved by this patch involves a leak of HTTP handler threads. The problem was that step 2b would successfully retrieve a stasis_app_control structure. Then step 2a would exit the channel from the event loop due to a hangup. Steps 3a and 4a would execute, and then finally steps 3b and 4b would. The problem is that at step 4b, when attempting to add a channel to a bridge, the thread would block forever since the channel would never execute the queued command since it was finished with the event loop. This meant that the HTTP handling thread would be leaked, along with any references that thread may have owned (in my case, I was seeing bridges leaked). The fix for this is to hone in better on when the channel has exited the event loop. The stasis_app_control structure has an is_done field that is now set at each point where the channel may exit the event loop. If step 2b retrieves a valid stasis_app_control structure but the control is marked as done, then the attempted operation exits immediately since there will be nothing to service the attempted command. ASTERISK-25091 #close Reported by Ilya Trikoz Change-Id: If66265b73b4c9f8f58599124d777fedc54576628	10 years ago
Joshua Colp	9668a1acb5	res_sorcery_memory_cache: Remove 'prefetch' option. ... To prevent confusion I am removing the prefetch option until such time as it is implemented. All other functionality, however, has been implemented. ASTERISK-25067 Change-Id: I9ce6aa3e5c6c5bc3c5baa8ff90fa036d73939895	10 years ago
Matt Jordan	8c0b917032	Merge "Parking: Add documentation for AMI ParkedCallSwap event."	10 years ago
Mark Michelson	59552c2d08	Parking: Add documentation for AMI ParkedCallSwap event. ... This event was added some time ago in order to clarify when a channel took the place of another channel in a parking lot. However, there was no XML documentation added for the event. This patch adds the XML documentation. ASTERISK-24900 #close Reported by Rusty Newton Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac	10 years ago
Joshua Colp	7cdb858651	Merge "res_pjsip: Add option to force G.726 to be treated as AAL2 packed."	10 years ago
Corey Farrell	ea9d5f155e	func_pjsip_aor: Fix leaked contact from iterator. ... ASTERISK-25162 #close Change-Id: Id79aa3c6fe490016ee98efc97ac4c1d3f461f97e	10 years ago
Kevin Harwell	93ac45d3bd	res_pjsip: Add option to force G.726 to be treated as AAL2 packed. ... Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615	10 years ago
mjordan	15c2208701	main/cdr: Carry over the disable flag when 'disable all' is specified ... The CDR_PROP function (as well as the NoCDR application) set the 'disable all' flag (AST_CDR_FLAG_DISABLE_ALL) on the current CDR. This flag is supposed to be applied to all CDRs that are currently in the chain, as well as all CDRs that may be created in the future. Currently, however, the flag is only applied to the existing CDRs in the chain; new CDRs do not receive the 'disable all' flag. In particular, this affects parallel dials, which generate new CDRs for each pair of channels in the dial attempt. This patch carries over the 'disable all' flag when it is specified on a CDR and a new CDR is generated for the chain. ASTERISK-24344 #close Change-Id: I91a0f0031e4d147bdf8a68ecd08304d506fb6a0e	10 years ago
Matt Jordan	b8bc15286f	main/cdr: Copy context/exten on chained CDRs for parallel dials in subroutines ... When a parallel dial occurs, a new CDR will be created for each dial attempt that is made. In most circumstances, the act of creating each CDR in the chain will include a step that updates the Party A snapshot, which causes the context/extension of the Party A to be copied onto the CDR object. However, when the Party A is in a subroutine, we explicitly do *not* copy the context/extension onto the CDR. This prevents the Macro or GoSub routine name from blowing away the context/extension that the channel was originally executing in. For the original CDR, this is not a problem: the original CDR already recorded the last known 'good' state of the channel just prior to it going into the subroutine. However, for newly generated CDRs in a chain, there is no context/extension set on them. Since we are in a subroutine, we will never set the Party A's context/extension on the CDR, and we end up with a CDR with no destination recorded on it. This patch updates the creation of a chained CDR such that it copies over the original CDR's context/extension. This is the last known "good" state of the CDR, and is a reasonable starting point for the newly generated CDR. In the case where we are not in a subroutine, subsequent code will update the location of the CDR from the Party A information; in the case where we are in a subroutine, the context/extension on the original CDR is the correct information. ASTERISK-24443 #close Change-Id: I6a3ef0d6e458d3b9b30572feaec70f2964f3bc2a	10 years ago
Matt Jordan	56d6b52a3f	Merge "bridge: When performing a blonde transfer update connected line information."	10 years ago
Mark Michelson	0b1a25054f	Merge "chan_sip.c: Update dialog fromtag after request with auth"	10 years ago
Mark Michelson	b9587fa9a8	Merge "app_directory: Fix crash when using the alias option 'a'."	10 years ago
Damian Ivereigh	19f60d9412	chan_sip.c: Update dialog fromtag after request with auth ... If a client sends and INVITE which is 401 rejected, then subsequently sends a new INVITE with the auth info and uses a different fromtag from the first INVITE, Asterisk will accept the new INVITE as part of the original dialog - match_req_to_dialog() specifically ignores the fromtag. However it does not update the stored dialog with the new fromtag. This results in Asterisk being unable to match future packets that are part of this dialog (such as the ACK to the OK or the OK to the BYE), and the call is dropped. This problem was originally found when using an NEC-i SV8100-GE (NEC SIP Card). * After a successful match of a packet to the dialog, if the packet is not a SIP_RESPONSE, authentication is present and the fromtags are different, the stored fromtag is updated with the one from the recent INVITE. ASTERISK-25154 #close Reported by: Damian Ivereigh Tested by: Damian Ivereigh Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e	10 years ago
Matt Jordan	bb00b26f35	chan_pjsip: Set the context and extension on the channel when created ... Prior to this patch, chan_pjsip was failing to pass the endpoint's context and the desired extension to the ast_channel_alloc_* routine. This caused a new channel snapshot to be issued without a context and extension, which can cause some reporting issues for users of AMI, CEL, and other APIs. The channel driver would later set the context and extension on the channel such that the channel would start in the correct location in the dialplan, but the information reported in the initial event would be incorrect. This patch modifies the channel driver such that it now passes the context and extension directly into the allocation routine. This provides the information in the new channel snapshot published over Stasis. ASTERISK-25156 #close Reported by: cloos Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e	10 years ago
Matt Jordan	a6446332fc	Merge "install_prereq: Check if is installed aptitude otherwise to install."	10 years ago
Joshua Colp	7230ee2efe	bridge: When performing a blonde transfer update connected line information. ... When performing a blonde transfer the code uses the old masquerade mechanism to move a channel around. As a result of this certain information, such as connected line, is moved between the channels involved. Upon completion of the move a frame is queued which is supposed to update the connected line information on the channel. This does not occur as the code considers it a redundant update since the masquerade operation updated the channel (but did not inform it of the new connected line information). The code also does not queue a connected line update to be handled by the thread handling the channel. Without this any other channel that may be loosely involved does not know it is talking to a different caller. This change does the following to resolve this: 1. The indicated connected line information is cleared upon completion of the masquerade operation when doing a blonde transfer. This prevents the connected line update from being considered redundant. 2. A connected line update frame is now queued upon the completion of the masquerade operation so any other channel loosely involved knows that there is a different caller. ASTERISK-25157 #close Reported by: Joshua Colp Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20	10 years ago
Richard Mudgett	a657ab12f9	app_directory: Fix crash when using the alias option 'a'. ... The voicemail.conf mailbox key/value pair is defined as: <mailbox>=[<password>[,<full-name>[,<email>[,<pager>[,<options>]]]]] Where all fields in the value including the field values are optional. Since the parsing code for the mailbox key/value pair is sloppy, this patch tightens the parsing for the directory information. * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options' respectively in search_directory_sub(). Those names make more sense. * Made sure that search_directory_sub() is dealing with the voicemail.conf mailbox options field if it even exists when looking for the 'hidefromdir' and 'alias' options. * Fix crash if a voicemail.conf mailbox is just <mailbox>=<password>,<name> when the 'a' option is used. If there were no fields after the name then the 'options' pointer was not checked for NULL. * Fix users.conf alias processing if the 'a' option is used. The wrong variable was used. ASTERISK-25087 #close Reported by: Chet Stevens Change-Id: I86052ea77307beddddba5279824d39dc0d593374	10 years ago
Richard Mudgett	30cd559345	DNS: Need to use the same serializer for a pjproject SIP transaction. ... All send/receive processing for a SIP transaction needs to be done under the same threadpool serializer to prevent reentrancy problems inside pjproject when using an external DNS resolver to process messages for the transaction. * Add threadpool API call to get the current serializer associated with the worker thread. * Pick a serializer from a pool of default serializers if the caller of res_pjsip.c:ast_sip_push_task() does not provide one. This is a simple way to ensure that all outgoing SIP request messages are processed under a serializer. Otherwise, any place where a pushed task is done that would result in an outgoing out-of-dialog request would need to be modified to supply a serializer. Serializers from the default serializer pool are picked in a round robin sequence for simplicity. A side effect is that the default serializer pool will limit the growth of the thread pool from random tasks. This is not necessarily a bad thing. * Made pjsip_resolver.c use the requesting thread's serializer to execute the async callback. * Made pjsip_distributor.c save the thread's serializer name on the outgoing request tdata struct so the response can be processed under the same serializer. ASTERISK-25115 #close Reported by: John Bigelow Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a	10 years ago
Richard Mudgett	b23f33e7e5	DNS: Fix some corner cases. ... * Fix query_set destruction before we are done kicking the queries off. * Fixed no queries requested handling. * Add empty queries request unit test. * Added missing allocation check in ast_dns_query_set_add(). * Made initial pjsip resolving query vector slightly larger. ASTERISK-25115 Reported by: John Bigelow Change-Id: Ie8be8347d0992e93946d72b6e7b1299727b038f2	10 years ago
Richard Mudgett	ae589da466	DNS: Remove trailing newline from summary and descriptions. ... Those trailing newlines mess up test formatting. Change-Id: I5e3f3a55b82c9d7acb9661201d4993d1958f1185	10 years ago
Richard Mudgett	83bc9d366d	pjsip_resolver.c: Fix debug code to only execute at acceptable debug level. ... Change-Id: I1716c93d6e097ad28128ecb9e806aac7a4180c8a	10 years ago
Richard Mudgett	6d49dccd85	DNS: Fix doxygen comments. ... Change-Id: Icafea3fb4ea64ac027561b23cbfe2b17997dc549	10 years ago
Richard Mudgett	b705c09dbb	res_pjsip.h: Fix some doxygen comments. ... Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976	10 years ago
Richard Mudgett	aa8479778e	taskprocessor.c: Remove extra unref from off-nominal path. ... Change-Id: Iee3bd8c8a528776056972066698fe735f0f6cf60	10 years ago
Mark Michelson	339f965cbe	Merge "chan_iax2: Prevent deadlock between hangup and sending lagrq/ping"	10 years ago
Mark Michelson	38c50df497	Merge "weakref attribute detection broken with gcc 4.6 and higher"	10 years ago
Mark Michelson	f8359cf1ee	Merge "res_pjsip_transport_websocket: Fix use-after-free bugs."	10 years ago
Ivan Poddubny	07f5f45e5a	res_pjsip_transport_websocket: Fix use-after-free bugs. ... This patch fixes use-after-free bugs caught by AddressSanitizer. 1. PJSIP transport manager may decide to destroy transport on its own. For example, when the contact registered via websocket has not renewed its registration in time. The transport was destoyed, but the websocket listener thread was still active until the socket closes, and then tried to call transport_shutdown on transport that has been freed. Also, the transport destructor accessed wstransport->rdata.tp_info.pool right after freeing memory that contained wstransport itself. This patch converts transport to an ao2 object, allowing it to be refcounted, so that it is available until both websocket listener and pjsip transport manager are finished with it. 2. The websocket listener deletes the last reference on websocket session when the tcp connection is closed, and it gets destroyed, but the transport manager may still use it, for example when disconnect happens in the middle of a SIP transaction. A new reference to websocket session has been added that is released with the transport to prevent this. ASTERISK-25096 #close Reported by: Josh Kitchens ASTERISK-24963 #close Reported by: Badalian Vyacheslav Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b	10 years ago
ibercom	f897f36721	weakref attribute detection broken with gcc 4.6 and higher ... GCC 4.7 Manual: http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html weakref ("target") A weak reference is an alias that does not by itself require a definition to be given for the target symbol. ASTERISK-22559 #close Reported by: Ibercom Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf	10 years ago
Matt Jordan	bbeb753e5e	Merge "Fix unsafe uses of ast_context pointers."	10 years ago
Corey Farrell	80621ce3c5	Fix unsafe uses of ast_context pointers. ... Although ast_context_find, ast_context_find_or_create and ast_context_destroy perform locking of the contexts table, any context pointer can become invalid at any time that the contexts table is unlocked. This change adds locking around all complete operations involving these functions. Places where ast_context_find was followed by ast_context_destroy have been replaced with calls ast_context_destroy_by_name. ASTERISK-25094 #close Reported by: Corey Farrell Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa	10 years ago
Kevin Harwell	53c1126090	AMI: Escape string values. ... So this issue is a bit complicated. Since it is possible to pass values to AMI that contain a '\r\n' (or other similar sequences) these values need to be escaped. One way to solve this is to escape the values and then pass the escaped values to the AMI variable parameter string building function. However, this puts the onus on the pre-build function to escape all string values. This potentially requires a fair amount of changes along with a lot of string allocations/freeing for all values. Surely there is a way to push this complexity down a level into the string building function itself? This of course is possible, but ends up requiring a way to distinguish between strings that need to be escaped and those that don't. The best way to handle this is by introducing a new format specifier in the format string. For instance a %s (no escape) and %S (escape). However, that is a bit weird and unexpected. So faced with those possibilities this patch implements a limited version of the first option. Instead of attempting to escape all string values this patch only escapes those values that make sense. This approach limits the number of changes and doesn't suffer from the odd format specifier problem. ASTERISK-24934 #close Reported by: warren smith Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0	10 years ago
Matt Jordan	8785d0ccbf	Merge "test_sorcery_memory_cache_thrash: Add unit tests for thrashing the memory cache."	10 years ago
Matt Jordan	5788c6db67	Merge "res_sorcery_memory_cache: Implement expire_on_reload option."	10 years ago
David M. Lee	9fca378b36	Fixes for OS X ... * Add some type casting so tv_usec can really be a long, instead of some strange platform specific type. * Add some .dylib style files to .gitignore. * Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer versions of GCC, when compiling the Homebrew formula for Asterisk, are not properly passing the -Xlinker options to the linker. Given that -Wl, does exactly the [same thing][], and does it properly, this patch changes the -Xlinker options to use -Wl, instead. [reasons unknown]: http://bit.ly/1SUbEYx [same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd	10 years ago
ibercom	d463bac574	CLI: Cosmetic issue - core show uptime ... Show uptime information ends with an unnecessary space. Now NEEDCOMMA is better defined. Change-Id: I11b360504a0703309ff51772ff8f672287f3c5a1	10 years ago
Joshua Colp	128fe4cee8	res_sorcery_memory_cache: Implement expire_on_reload option. ... This change implements the expire_on_reload option for memory caches. If enabled and a reload is performed all objects within the cache will be expired and the cache emptied. ASTERISK-25067 Reported by: Matt Jordan Change-Id: Id46aa1957d660556700e689e195eed57c989b85e	10 years ago
Joshua Colp	028edae82e	test_sorcery_memory_cache_thrash: Add unit tests for thrashing the memory cache. ... This change adds a CLI command which can perform memory cache thrashing as well as unit tests which perform thrashing under the following configurations: 1. Low number of unique objects that go stale after 1 second 2. Low number of unique objects that expire after 1 second 3. Low number of unique objects which are constantly updated 4. Large number of unique objects which exceed a defined cache size 5. Large number of unique objects which exceed a defined cache size that also expire and go stale rapidly 6. Large number of unique objects which expire and go stale rapidly 7. Large number of unique objects For all of the above there are a large number of threads constantly attempting to retrieve random objects and each test runs for a few seconds. ASTERISK-25067 Reported by: Matt Jordan Change-Id: I8c8ceff977332c80ed4a31f10d694d48552b2f78	10 years ago
Mark Michelson	86c79314f1	Merge "res_sorcery_memory_cache: Add test event when a refresh occurs."	10 years ago
Matt Jordan	269fbff366	Merge "Remove const cast from leaf functions."	10 years ago
Joshua Colp	19de2bbc5f	res_sorcery_memory_cache: Add test event when a refresh occurs. ... This change adds a testsuite event for when a refresh occurs. This is useful as it provides a guaranteed mechanism of knowing when it has occurred instead of waiting an arbitrary amount of time. ASTERISK-25067 Reported by: Matt Jordan Change-Id: Iaa6b8d2d6bab7f99ee08e1c8908b8272a8987e65	10 years ago
Rodrigo Ramírez Norambuena	6737ded058	install_prereq: Check if is installed aptitude otherwise to install. ... If in Debian or system based, dont have aptitude installed the script do nothing. This patch checked if aptitude installed, if not installed. Also, if execute script with all packages installed yet, the script not show nothing and return exit 1 because the command 'grep' get nothing from pipe from 'awk'. ASTERISK-25113 #close Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f	10 years ago
Mark Michelson	92ccffd9e6	res_pjsip: Prevent access of NULL channels. ... It is possible to receive incoming requests or responses after the channel on an ast_sip_session has been destroyed and NULLed out. Handlers of these sorts of requests or responses need to be prepared for the possibility that the channel is NULL or else they could cause a crash. While several places have been amended to deal with NULL channels, there were still a couple of places that needed updating. res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to return early if there is no channel on the session. res_pjsip_session.c: When handling a 302 response, we need to stop the redirecting attempt if there is no channel on the session. ASTERISK-25148 #close reported by Mark Michelson Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9	10 years ago
George Joseph	d355ee7ff3	res_pjsip/location: Fix ref leak in contact_apply_handler ... contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status to force the creation of a contact_status object whenever a new contact is added but it didn't unref the returned object. Added an ao2_cleanup(status) to plug the leak. ASTERISK-25141 Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40 Reported-by: Corey Farrell	10 years ago
Richard Mudgett	6d8dc9bb5c	res_pjsip: Remove outgoing authentication code no longer needed. ... Associated with ASTERISK-25131 Change-Id: Iefa3b2066cfd8b108a90d2dd4a64d92c3a195d33	10 years ago
Richard Mudgett	00a47ffc7e	res_pjsip_session: Fix cherry pick to master compile error. ... ASTERISK-25131 Reported by: Richard Mudgett Change-Id: I87c9c96ae4a8fe2bc8a0ddea6958a2ad9cefd8e3	10 years ago
Joerg Sonnenberger	9472bbaa95	Remove const cast from leaf functions. ... app_control_register_rule and app_control_unregister_rule lock/unlock the queue, which is a mutating operation according to the ao2_lock/_unlock prototype. Depending on the specific (implicit) casts in SCOPED_LOCK and RAII_VAR, the compiler may warn or not. As the only callers of those functions do not have the const, get consistent results by just dropping it. Change-Id: Ib9e6296155a39bc5d627142a3828180c3cfe8fbb	10 years ago