asterisk

Commit Graph

Author	SHA1	Message	Date
Jonathan Rose	729c91b312	Don't attempt free of NULL ptr in pbx.c handle_presencechange (closes issue AST-921) Reported by: Guenther Kelleter Patches: nullptr.patch uploaded by Guenther Kelleter (license 6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370466 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	39ae307bdb	Silence a warning message from older versions of GCC. Revision 370426 introduced the use of a nested function in tests/test_acl.c, but the lack of the 'auto' scope specifier on the function and a forward declaration resulted in compilation errors on the automated test systems. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370453 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tzafrir Cohen	6f8bb47833	chan_oss: fix "sample rate" error message Merged revisions 370428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 Merged revisions 370432 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370433 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	af3ef19d00	Rewrite a comment that didn't adequately explain the code it was documenting. ........ Merged revisions 370429 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370430 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370431 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	b74acabdc7	Update CHANGES for list/negation ACL feature. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	0385c0e9cb	Allow permit/deny ACL lines to contain multiple items and negated entries. Rules in ACLs (specified using 'permit' and 'deny') can now contain multiple items (separated by commas), and items in the rule can be negated by prefixing them with '!'. This simplifies Asterisk Realtime configurations, since it is no longer necessray to control the order that the 'permit' and 'deny' columns are returned from queries. Review: https://reviewboard.asterisk.org/r/1592/ Initial patch contributed by Tilghman Lesher Unit tests written by Kevin P. Fleming git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	190d130cbe	Build is underway so logging can go away. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370420 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	0ef30a9071	Temporarily enable pj logging to console for debugging pjnath issue exposed by build slave. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370419 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	8eaba809ab	Remove code, that operate with cdr in attempt_transfer(). That was removed somewhere between 1.2 and 1.4 and acidentaly put back in chan_unistim.	13 years ago
Kevin P. Fleming	b5193428a7	Enable usage of system-provided iLBC library. The WebRTC version of the iLBC codec is now package as a library and is available on some platforms. This patch allows codec_ilbc to be built against that library if it is present. Review: https://reviewboard.asterisk.org/r/1964/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b6a0ae0b35	Unit tests for the Jitter Buffer API; remove unnecessary resync This patch includes the following: * Unit tests for the abstract Jitter Buffer API. This includes both fixed and adaptive flavors, testing nominal creation, frame input, frame retrieval, resyncing; off nominal frame input overflow, out of order, and others. * Tweaks to the abstract_jb API to remove the unnecessary resync_threshold parameter from the create function (resync_threshold is already in the struct passed into the create function) * Ensure the fixed jitter buffer is empty before destroying it, to avoid an ASSERT * Don't "resync" the adaptive jitter buffer. The mechanism that was being used actually causes the jitter buffer to think its being overflowed by going around the jitterbuf API and attempting to 'resynch' it improperly. If a resync is needed, the jitter buffer will do it properly by itself. Note that this is only an optimization needed for trunk, as the worst that happens is the loss of three voice packets before the adaptive jitter buffer will resync anyway. Review: https://reviewboard.asterisk.org/r/2035 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a28e6fc7bd	Add separate configuration options for subscription and registration minexpiry and maxexpiry. This offers more fine-grained control over how long subscriptions last without negatively affecting the expiration range for registrations. Uploaded by: Guenther Kelleter(license #6372) Review: https://reviewboard.asterisk.org/r/2051 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	ec14c2563e	Improve documentation for the SHELL() dialplan function. ........ Merged revisions 370383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370384 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	014e8a0a80	Add notes to UPGRADE.txt about addition of msg_id to VoiceMails. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	4f8fd4f435	Blocked revisions 370361 ........ Free any datastores attached to dummy channels. Revision 370205 added the use of a datastore attached to a dummy channel to resolve a memory leak, but ast_dummy_channel_destructor() in this branch did not free datastores, resulting in a continued (but slightly smaller) memory leak. This patch backports the change to free said datastores from the Asterisk trunk. (related to issue AST-916) ........ Merged revisions 370360 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	941941ce98	Update UPGRADE.txt with notes about ICE support and res_xmpp. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	92abf49458	Update CHANGES for Asterisk 11 This updates the CHANGES file with things that were committed for Asterisk 11, but were not noted in that file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370353 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	4d6b524b61	Prevent multiple local candidates from being added with the same information and add support for disabling ICE on a per-peer basis. (closes issue ASTERISK-20088) Reported by: wimpy Review: https://reviewboard.asterisk.org/r/2044/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	13427db64c	Fix segfault introduced by conversion to ACO API The value "none" is specified in the config file as a valid value for the "video_mode" option. The code prior to the ACO conversion did not check for "none", but just ignored it and relied on the default zero value. The parsing with ACO is more strict, so without handling "none" specifically, parsing would fail. When parsing failed, but the module loaded anyway, the config info would never be stored, and one place in the code did not check for this case and would segfault. It was also possible that the aco_info struct's internals would be destroyed and used as well. This patch keeps the module from loading after parse failures, adds the "none" option to "video_mode", registers CLI functions only after parsing has completed, checks the config data for NULL before accessing it, and returns -1 on some allocation failures when initializing. (closes issue ASTERISK-20159) Reported by: Birger "WIMPy" Harzenetter Tested by: Birger "WIMPy" Harzenetter Patches: confbridge_fix3.txt uploaded by Terry Wilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	a5e10001b2	chan_iax2: Fix a segfault introduced by call ID logging Didn't previously check that a non NULL IAX channel was stored in the array at the requested position before attempting iax_pvt_callid_get (closes issue ASTERISK-20145) Reported by: Birger "WIMPy" Harzenetter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	fbf4040a36	Clean up ManagerEvent Dial documentation The paragraph describing the SubEvent belongs with the SubEvent parameter itself, and not with its enum values. The order of parsing was placing the description after the last enum, which isn't correct. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370329 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	c2d9192660	Fix build error in chan_misdn from commit 370316 chan_misdn was not updated properly to account for a change in parameters for HANGUPCAUSE functionality. It now builds properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	afaa23864b	Export the ast_websocket_set_nonblock function for use by other modules. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	cb9756daa2	Add hangupcause translation support The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better facilitate access to the AST_CAUSE translations for technology-specific cause codes. The HangupCauseClear application has also been added to remove this data from the channel. (closes issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	499a445af2	Update CHANGES about adding the AccountCode header to the AMI Hangup event. (issue ASTERISK-19963) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370315 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	54991ca2a7	Add the AccountCode header to the AMI Hangup event. It's harder to correlate the Newchannel and Hangup AMI events without specifying "AccountCode" in both. (closes issue ASTERISK-19963) Reported by: Oleg A. Arkhangelsky Patches: hangup_acctcode.diff (license #6397) patch uploaded by Oleg A. Arkhangelsky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	2f674bcdd1	Convert app_confbridge to use the config options framework Review: https://reviewboard.asterisk.org/r/2024/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b78fd0ac89	Fix compiler warnings. gcc (GCC) 4.2.4 has problems casting away constness. ........ Merged revisions 370275 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370277 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	86ff5585fd	Add the ability to specify technology specific documentation A number of applications/AMI commands in Asterisk have specific behavioral differences depending on the resource or channel technology those applications are executed on. For example, the MessageSend application/ command is technology agnostic, but how the channel drivers that support that functionality behave is dependant on the protocols and channel driver implementation. Prior to this patch, those details were either documented in the application/command documentation itself, or were left undocumented. This patch adds a new element to the documentation schema, <info/>. An info node is essentially a piece of technology specific reference information that can be included by any top level XML documentation node. For example, the MessageSend application can now include XMPP/SIP specific information, where that technology specific information can be defined in chan_motif/res_xmpp/ chan_sip. Likewise, that information can also be included in the MessageSend AMI command. Review: https://reviewboard.asterisk.org/r/2049 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f802787924	Fix compilation error when MALLOC_DEBUG is enabled To fix a memory leak in CEL, a channel datastore was introduced whose destruction function pointer was pointed to the ast_free macro. Without MALLOC_DEBUG enabled this compiles as fine, as ast_free is defined as free. With MALLOC_DEBUG enabled, however, ast_free takes on a definition from a different place then utils.h, and became undefined. This patch resolves this by using a reference to ast_free_ptr. When MALLOC_DEBUG is enabled, this calls ast_free; when MALLOC_DEBUG is not enabled, this is defined to be ast_free, which is defined to be free. (issue AST-916) Reported by: Thomas Arimont ........ Merged revisions 370273 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370274 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	245f6538e7	Handle extremely out of order RFC 2833 DTMF The current implementation of RFC 2833 DTMF handling in res_rtp_asterisk will, if a packet arrives out of order, drop the packet. This is to prevent duplicate ton generation in the Asterisk core. Since the RTP layer does not buffer data itself, this is the only option the RTP layer currently has for handling packets that arrive out of order. For the most part, this doesn't matter. For a particular digit, so long as a BEGIN packet arrives before the first END packet, the digit will be produced. If subsequent BEGIN packets arrive interleaved with the ENDs, they will be dropped; likewise, if the BEGIN or END packets themselves are out of order, those packets are dropped but sufficient information is conveyed to the Asterisk core to produce the appropriate digit. For certain sequences of DTMF packets - most notably when, for a particular digit, an END packet arrives before any BEGIN packet for that digit - this is a real problem. When an END arrives before any BEGINs, the END packet is dropped - but at the same time, it causes subsequent BEGIN packets for that digit to be ignored. When the next in order END packet arrives, it too is dropped - Asterisk believes that there was no initial BEGIN. The solution this patch provides is to trust the END packet to convey the information needed for the Asterisk core to produce the DTMF digit. If we receive an END packet, and it: * Has a timestamp greater then the last timestamp received from an END packet * Does not have the same sequence number as the last received sequence number (and is thus not an END packet retransmission) Then we send the END frame up to the Asterisk core. It contains enough DTMF information for Asterisk to produce the digit. On the other hand, if we receive a BEGIN or continuation packet that occurs with a timestamp equal to or less then the last END timestamp, then we've received something out of order - but we already have received enough information to produce the digit. These packets are dropped. Much thanks goes to Olle Johansson (oej) for providing the idea for this solution. Review: https://reviewboard.asterisk.org/r/2033/ (closes issue ASTERISK-18404) Reported by: Stephane Chazelas Tested by: Matt Jordan ........ Merged revisions 370252 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370271 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	ded09e3682	named_acl: Remove systemname option from acl.conf, use asterisk.conf value Review: https://reviewboard.asterisk.org/r/2057/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	d13e015784	CallID Logging: Remove new line/carriage return from callID change test event git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	3a2757923c	Use the bruteforce method to get debugging enabled for pjproject. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	bfa31f5676	Turn on debugging for pjproject so we can get a better idea of what is causing the generic CCSS test crash. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	5e4ee6076c	callid logging: Issue test events when the callid is changed for a channel Review: https://reviewboard.asterisk.org/r/2054/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	4a4189b085	Resolve severe memory leak in CEL logging modules. A customer reported a significant memory leak using Asterisk 1.8. They have tracked it down to ast_cel_fabricate_channel_from_event() in main/cel.c, which is called by both in-tree CEL logging modules (cel_custom.c and cel_sqlite3_custom.c) for each and every CEL event that they log. The cause was an incorrect assumption about how data attached to an ast_channel would be handled when the channel is destroyed; the data is now stored in a datastore attached to the channel, which is destroyed along with the channel at the proper time. (closes issue AST-916) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2053/ ........ Merged revisions 370205 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370206 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	79087cbbd5	Ensure that all ast_datastore_info structures are 'const'. While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370187 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	8401e81383	Fix a crash in pjnath when starting an ICE connectivity check and immediately destroying the ICE session. The initial ICE connectivity check is scheduled as a timer item that is to be executed immediately. It is possible for this timer item to start executing while the ICE session it is working on is destroyed. To reduce the chance of this any timer items that need to be immediately executed will be executed within the thread that has started the initial ICE connectivity check. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	cbdb2dbb0e	Fix a crash occurring as a result of excess stack usage. This fix involves moving the allocation of some temporary codec structures to the heap and also reduces the number of maximum payloads to something more sane for both regular and low memory builds. (closes issue ASTERISK-20140) Reported by: jonnt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370171 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	9278b5e51e	Added option 'interdigit_timer' to unistim.conf to make able controll hardcoded dial timeout constant.	13 years ago
Joshua Colp	cd91570bc6	Add pubsub unsubscription support so subscriptions do not linger for MWI and device state progatation. The pubsub code did not attempt to remove subscriptions at all. This has now changed so that if a client is being disconnected it will unsubscribe. It will also unsubscribe at connection time so if it unexpectedly disconnected duplicate subscriptions will not occur. (closes issue ASTERISK-19882) Reported by: mattvryan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	44345b0973	Fix a crash as a result of propagating MWI or device state over XMPP when the client is disconnected. The MWI and device state propagation code wrongly assumes that an XMPP client connection will remain established at all times. This fix corrects that by making the lifetime of the subscription the same as the lifetime of the connection itself. As the connection is established and disconnected the subscription itself is created and destroyed. (closes issue ASTERISK-18078) Reported by: elguero git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370152 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	6027b26fa7	Code cleanup and bugfix in chan_sip outboundproxy parsing. The bug was clearing the global outboundproxy when a peer-specific outboundproxy was bad. The cleanup reduces duplicate code. Review: https://reviewboard.asterisk.org/r/2034/ Reviewed by: Mark Michelson ........ Merged revisions 370131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370132 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	fdd39eae58	Fix an issue where a service discovery request could crash Asterisk. A server sending a service discovery request to us may or may not put a from attribute in the message. If the from attribute is present use it in the to attribute for the result. If the from attribute is not present do not add a to attribute. (issue ASTERISK-16203) Reported by: wubbla git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370126 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	3b59ab1c77	Fix a bug where some XMPP servers would reject authentication. We need to use the user portion of the JID and not the full configured username. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	7a78aa39d1	Add missing namespace for old non-SASL based authentication. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	f234eae9ee	Fix a bug exposed by the testsuite where text streams would no longer be parsed correctly. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	25e721ee9b	Add comments about the BUILD_NATIVE change This is a significant change and mention of it should have gone into UPGRADE.txt and CHANGES. ........ Merged revisions 370081 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370082 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	5d20f60337	Fix an issue where specifying the resource in the username would cause authentication to fail. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370073 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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