asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	ec37b8e527	Part of the merge did not happen correctly, which resulted in a compile error git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a6adc84e69	SIP option to specify outbound TLS/SSL client protocol. chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip. (closes issue #14770) Reported by: TheOldSaint (closes issue #14768) Reported by: TheOldSaint Review: http://reviewboard.digium.com/r/240/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	d35fd35ae3	Outgoing PTP redirected calls did not wait for the COLR from the redirected-to party. For outgoing PTP redirected calls, you now need to use the inhibit(i) option on all of the REDIRECTING statements before dialing the redirected-to party. You still have to set the REDIRECTING(to-xxx,i) and the REDIRECTING(from-xxx,i) values. The PTP call will update the redirecting-to presentation when it becomes available and queue the redirecting update to the calling channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a866a75900	Merge str_substitution branch. This branch adds additional methods to dialplan functions, whereby the result buffers are now dynamic buffers, which can be expanded to the size of any result. No longer are variable substitutions limited to 4095 bytes of data. In addition, the common case of needing buffers much smaller than that will enable substitution to only take up the amount of memory actually needed. The existing variable substitution routines are still available, but users of those API calls should transition to using the dynamic-buffer APIs. Reviewboard: http://reviewboard.digium.com/r/174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	0ea83eab48	Removing crufty code that is no longer necessary. Code cleanup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Brooks	7f5909b23f	Blocked revisions 191096 via svnmerge ........ r191096 \| dbrooks \| 2009-04-29 13:07:59 -0500 (Wed, 29 Apr 2009) \| 8 lines Patch to fix tab-completion crash on "remove extension" This patch simply removes some old code back before Asterisk used editline. This fixes the crash that occurred when tab-completing "remove extension". (closes issue #14689) Reported by: isaacgal ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	ca138fc807	Consistent SSL/TLS options across conf files ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though. Review: http://reviewboard.digium.com/r/237/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	77f08759b5	Log an error message if indications.conf is not found. (closes issue #14990) Reported by: tzafrir Patches: indications_err.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	1e016da893	Fix app_queue XML documentation. I think it would behoove us to force "make validate-docs" to be run after the XML documentation has been generated if dev-mode is enabled. (closes issue #14989) Reported by: tzafrir Patches: app_queue_xml.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2c1ffef923	Resolve Solaris build issues and add some API documentation. (issue #14981) Reported by: snuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	a082ad616f	Add support setting CPC from channel variable git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	6389ec15fb	Make sure that we do not clear the down flag on the BRI during PTMP link transients git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	4bf441ca39	UniqueID column has a maximum size of 150 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f321d08113	Build XML documention from only the source files that have docs in them Change the build process so that doc/core-en_US.xml is dependent solely on the source files that have documentation in them, not on all source files. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	97e5d179ca	Remove Makefile rules for bison and flex sources We never, ever want these files to processed automatically, because we store the output files in Subversion and users should never need to rebuild them. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
TransNexus OSP Development	2e0f8bcbc8	Updated for OSP Toolkit 3.5. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190830 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	fb030f24ef	Fix a small memory leak on error in ast_channel_alloc(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	89d06c7759	Make PTP DivertingLegInformation3 message behavior closer to the specifications. * Wait for a DivertingLegInformation3 message after receiving a DivertingLegInformation1 message to complete the redirecting-to information before queuing a redirecting update to the other channel. * A DivertingLegInformation2 message should be responded to with a DivertingLegInformation3 when the COLR is determined. If the call could or does experience another redirection, you should manually determine the COLR to send to the switch by setting REDIRECTING(to-pres) to the COLR and setting REDIRECTING(to-num) = ${EXTEN}. * A DivertingLegInformation2 message must have an original called number if the redirection count is greater than one. Since Asterisk does not keep track of this information, we can only indicate that the number is not available due to interworking. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b88343b248	Don't warn on pipe in the System call. (closes issue #14979) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190726 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	a9657a86bf	Merged revisions 190721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190721 \| kpfleming \| 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) \| 7 lines Fix 'inconsistent line endings' when autoconf 2.63 is used Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190725 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	639ece2a31	Merged revisions 190661-190662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190661 \| russell \| 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) \| 9 lines Resolve a crash in res_smdi when used with chan_dahdi. When chan_dahdi goes to get an SMDI message, it provides no search criteria. It just grabs the next message that arrives. This code was written with the SMDI dialplan functions in mind, since that is now the preferred method of using SMDI. However, this broke support of it being used from chan_dahdi. (closes AST-212) ........ r190662 \| russell \| 2009-04-27 14:03:59 -0500 (Mon, 27 Apr 2009) \| 2 lines Fix a typo from 190661. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	1d941ad821	Allow for a position to be specified when entering a queue. This would allow for one to add a caller to a specific place in the queue instead of just placing the caller in the back every time. To help facilitate some interesting manipulations, a new channel variable called QUEUEPOSITION has been added. When a caller is removed from a queue, his position in that queue is stored in the QUEUEPOSITION variable. One such strategy an administrator can employ is to allow for the removal of a caller from one queue followed by the insertion of the same caller into a separate queue in the same position. Review: http://reviewboard.digium.com/r/189 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190626 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	09cde5a40c	Update warning message to not have pipes and contain all options. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190622 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	68a9f7aca1	Fix a bug where we tried to send events out when no sessions container was present. This commit stops a warning message (user_data is NULL) from getting output when manager events get sent before manager is initialized. This happens because manager is initialized after modules are loaded and the act of loading modules triggers manager events. (issue #14974) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190586 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	3b68be6aaa	Remove nonexistent option from sip.conf.sample. The option to choose which connected line header to use is not 'rpid_header' but 'sendrpid' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8f0b88c8c8	TLS/SSL private key option Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up. Review: http://reviewboard.digium.com/r/234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	c95c065903	There is no need to use the struct ast_party_connected_line.source update values. The messages sent by a technology when a connected line update is received are best determined by the current call state of the channel. The struct ast_party_connected_line.source value is really only useful as a possible tracing aid. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	014aa91b84	Update comment. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f052718a80	Add \since tag for new API calls. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	c878b21887	Fix a build error. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	e9e1c68df4	Blocked revisions 190187 via svnmerge	17 years ago
Russell Bryant	cba19c8a67	Convert the ast_channel data structure over to the astobj2 framework. There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	f314c5f13f	Fix nat setting on RTP instances. (closes issue #14827) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190421 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	bbcf30787d	Merged revisions 190356 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190356 \| russell \| 2009-04-23 16:07:07 -0500 (Thu, 23 Apr 2009) \| 2 lines Remove a bogus ast_channel_unlock(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	d6c48bc134	Labels are sometimes (most of the time?) NULL for extensions. (closes issue #14895) Reported by: chris-mac Patches: 20090423__bug14895__2.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	ce6ebaef97	Support HTTP digest authentication for the http manager interface. (closes issue #10961) Reported by: ys Patches: digest_auth_r148468_v5.diff uploaded by ys (license 281) SVN branch http://svn.digium.com/svn/asterisk/team/group/manager_http_auth Tested by: ys, twilson, tilghman Review: http://reviewboard.digium.com/r/223/ Reviewed by: tilghman,russellb,mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	80b8d34377	Merged revisions 190286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190286 \| file \| 2009-04-23 16:13:18 -0300 (Thu, 23 Apr 2009) \| 6 lines Fix a bug in chan_local glare hangup detection. If both sides of a Local channel were hung up at around the same time it was possible for one thread to destroy the local private structure and have the other thread immediately try to remove the already freed structure from the local channel list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8f81deab25	Fix reversed behavior of leavewhenempty option in queues.conf. (closes issue #14650) Reported by: alecdavis Patches: 14650.patch uploaded by mmichelson (license 60) Tested by: mmichelson, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	ada8ae56e1	Fix a double free issue with the Pickup dialplan application. As part of the pickup process the connected line information is updated. Part of this process does a shallow copy of the target channel's connected line information to a local structure. Once complete the structure contents are freed. As a result any information in the target channel's connected line information structure is no longer valid. This change will now set the contents back to a clean state so that the freeing of the target channel's connected line information structure when the channel is destroyed will no longer try to double free things. (closes issue #14839) Reported by: lmsteffan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	1ce1f1bb1f	Fix example that could fail in certain circumstances git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	25cea89d90	Merged revisions 190092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190092 \| tilghman \| 2009-04-22 16:35:03 -0500 (Wed, 22 Apr 2009) \| 7 lines Detect availability of pthread_rwlock_timedwrlock() before using it. (closes issue #14930) Reported by: tilghman Patches: 20090420__bug14930.diff.txt uploaded by tilghman (license 14) Tested by: mvanbaak, tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	11ac1f7e11	Fix building of chan_h323 with gcc-3.3 There seems to be a bug with old versions of g++ that doesn't allow a structure member to use the name list. Rename list member to group_list in ast_group_info and change the few places it is used. (closes issue #14790) Reported by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	3fb648d8fa	Add funcs for manipulating delimited lists in the dialplan Adds PUSH and POP for appending to and retrieving/removing from the end of a list and UNSHIFT and SHIFT for insert to and retrieiving/ removing from the beginning of a list. Review: http://reviewboard.digium.com/r/230 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190000 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	2f8da7106d	Make chan_h323 respect packetization settings and fix small reload issue. Previously, packetization settings were ignored and now they are not. A new config option 'autoframing' has been added to mirror the way chan_sip handles it. Turning on the autoframing option (available both as a global option or per peer) overrides the local settings with the remote packetization settings. Testing was performed with varying packetization levels with the following codecs: ulaw, alaw, gsm, and g729. Also, an unrelated config reload issue has been fixed in the case of the config file not changing. (closes issue #12415) Reported by: pj Patches: 2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7), modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	3b64f588e2	Blocked revisions 189991 via svnmerge ........ r189991 \| jpeeler \| 2009-04-22 14:20:53 -0500 (Wed, 22 Apr 2009) \| 15 lines Make chan_h323 respect packetization settings Previously, packetization settings were ignored and now they are not. A new config option 'autoframing' has been added to mirror the way chan_sip handles it. Turning on the autoframing option (available both as a global option or per peer) overrides the local settings with the remote packetization settings. Testing was performed with varying packetization levels with the following codecs: ulaw, alaw, gsm, and g729. (closes issue #12415) Reported by: pj Patches: 2009012200_h323packetization.diff.txt uploaded by mvanbaak (license 7), modified by me ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	559f908016	Fix call parking callback. Pipes -> Commas. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	9cbd90a864	Do not continue to receive DTMF, when the channel is hungup and about to be destroyed. (closes issue #14858) Reported by: barryf Patches: 20090421__bug14858.diff.txt uploaded by tilghman (license 14) Tested by: barryf git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	ba6dab1a1b	Merged revisions 189849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189849 \| mvanbaak \| 2009-04-22 16:29:28 +0200 (Wed, 22 Apr 2009) \| 12 lines replace sed with tr to remove \r from downloaded file On some systems, sed does not recognize \r in the pattern the way it was used here. Use tr instead because this works the same across systems. (closes issue #14936) Reported by: leobrown Patches: 2009042201_14936.diff.txt uploaded by mvanbaak (license 7) Tested by: leobrown, mvanbaak ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	07a115bead	Detect liblua on SuSE, and add libm for linking for Fedora. (Reported via the -dev list, Subject: Compiling Asterisk with LUA) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189813 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	00d7c4fefc	Fixes segfault when switching UDP to TCP in sip.conf after reload. If transport in sip.conf is switched from UDP to TCP, Asterisk segfaults right after issuing a sip reload. The problem is the socket type is changed to TCP but the fd may still be present for UDP. Later, when the TCP session should be created or set using an existing one, it isn't because the old file descriptor is still present. Now every time transport is changed during a sip.conf reload, the file descriptor is set to -1, signifying it must be created or found. (closes issue #14727) Reported by: pj Tested by: dvossel Review: http://reviewboard.digium.com/r/229/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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