7041 Commits (bcdf507f3c67ad8c01ef1d989ce53e45bea0f201)

Author	SHA1	Message	Date
Richard Mudgett	bcdf507f3c	Change D-channel warning to be less confusing on non-NFAS setups. ... The "No D-channels available! Using Primary channel as D-channel anyway!" WARNING message has been confusing on non-NFAS setups. The message refers to things that are NFAS specific. * Changed the warning to several different warnings to be more accurate for the situation and less confusing as a result: "No D-channels up! Switching selected D-channel from X to Y.", "No D-channels up!", and "D-channel is down!". git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	2adb133feb	Outbound SIP OPTIONS messages will now include fromuser of related peer. ... This behavior matches up more closely with the way invite/register/etc are handled. This patch also modifies some adjacent code for code style compliance. Pretty minor. (closes issue ASTERISK-17616) Reported by: Jeremy Kister Patches: chan_sip.c-options-fromuser-fix-v1.patch uploaded by Jeremy Kister (license #6232) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342061 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	d6f1839114	Outgoing calls with Google Voice ... Google has recently make some changes (again) to their protocol. Rather then patching asterisk to flip between the two different methods, we now allow both. Lets hope this keeps Google Voice happy for a while. (closes issue ASTERISK-18714) Reported by: Iordan Iordanov Patches: chan_gtalk.patch uploaded by Iordan Iordanov (licenses 6311) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	8eb030a3a2	Don't use is_int() since it doesn't link well on all platforms ... Just create an normal API function in strings.h that does the same thing just to be safe. ASTERISK-17146 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341379 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	eae454ca3f	Don't sent in-dialog requests like UPDATE when Asterisk has not yet received a Contact URI from a UAS ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	432657163f	Don't resolve numeric hosts or contact unresolved hosts ... If a SIP dial string contains a numeric hostname that is not a peer name, don't try to resolve it as it is unlikely that someone really means Dial(SIP/0.0.4.26) when Dial(SIP/1050) is called. Also, make sure that create_addr returns -1 if an address isn't resolved so that we don't attempt to send SIP requests to an address that doesn't resolve. (closes issue ASTERISK-17146, ASTERISK-17716) Review: https://reviewboard.asterisk.org/r/1532/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f2b371fedf	More parking issues. ... * Fix potential deadlocks in SIP and IAX blind transfer to parking. * Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the parkext_exclusive option with transfers (Park(,,,,,exclusive_lot) parameter). Created ast_park_call_exten() and ast_masq_park_call_exten() to maintian API compatibility. * Made masq_park_call() handle a failed ast_channel_masquerade() setup. * Reduced excessive struct parkeduser.peername[] size. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341254 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	2426e2604e	Initialize variables before calling parse_uri ... If parse_uri was called with an empty URI, some pointers would be modified and an invalid read could result. This patch avoids calling parse_uri with an empty contact uri when parsing REGISTER requests. AST-2011-012 (closes issue ASTERISK-18668) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	b951592017	Don't try to remove peers without IPs from peers_by_ip ... (closes issue ASTERISK-18696) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	0fa2f5914e	Quiet RTCP Receiver Reports during fax transmission ... RTCP is now disabled for "inactive" RTP audio streams during SIP T.38 sessions. The ability to disable RTCP streams in res_rtp_asterisk was missing, so this code was added to support the bug fix. (closes issue ASTERISK-18400) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340970 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	598b45b175	storing the route-set also on a 181 response not only on 180,182 or 183. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	eb38856434	Initialize ast_sockaddr before calling ast_sockaddr_resolve ... Avoid possible jump based on unitialized value git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	3bc7b5d2c9	Store route-set from provisional SIP responses so early-dialog requests can be routed properly ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	610a2997dd	Update SIP realtime fullcontact regardless of caching ... We should update the fullcontact field in the realtime table whether or not rtcachefriends is set. There is no reason to treat a non-cached realtime entity differently than a cached in this regard. (closes issue ASTERISK-18446) Reported by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0c069b5653	Initialize the PRI channel alarms properly on startup. ... The PRI channel alarms were initialized with an inverted sense. (closes issue ASTERISK-18710) Reported by: Tzafrir Cohen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	35fcb785af	Fix verbose messages when IPv6 logic was added ... (closes issue ASTERISK-18612) Reported by: Tim Osman git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340418 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	fbc51bb795	Add protection for SS7 channel allocation and better glare handling. ... * Added a CLI "ss7 show channels" command that might prove useful for future debugging. * Made the incoming SS7 channel event check and gripe message uniform. * Made sure that the DNID string for an incoming call is always initialized. (issue ASTERISK-17966) Reported by: Kenneth Van Velthoven Patches: jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a458ac621e	Fix some potential deadlocks pointed out by helgrind. ... * Fixed deadlock potential calling dialog_unlink_all() in __sip_autodestruct(). Found by helgrind. * Fixed deadlock potential in handle_request_invite() after calling sip_new(). Found by helgrind. * The sip_new() function now returns with the created channel already locked. * Removed the dead code that starts a PBX in in sip_new(). No sip_new() callers caused that code to be executed and it was a bad thing to do anyway. * Removed unused parameters and return value from dialog_unlink_all(). * Made dialog_unlink_all() and __sip_autodestruct() safely obtain the owner and private channel locks without a deadlock avoidance loop. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	21bb14654b	Updated chan_sip to place calls on hold if SDP address in INVITE is ANY ... This patch fixes the case where an INVITE is received with c=0.0.0.0 or ::. In this case, the call should be placed on hold. Previously, we checked for the address being null; this patch keeps that behavior but also checks for the ANY IP addresses. Review: https://reviewboard.asterisk.org/r/1504/ (closes issue ASTERISK-18086) Reported by: James Bottomley Tested by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	5e05620bb7	Fix compilation issue, caused by missed session structure ... (closes issue ASTERISK-18694) Reported by: alex70 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@339938 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	7d3b4d5e80	Fix segfault in Unistim channel ...	14 years ago
Igor Goncharovskiy	6d6ed815cc	Fix char array cast as short array in send_client() function (for ARM ...	14 years ago
Richard Mudgett	06e6b7bba1	Fix debugging messages generated by 'udptl debug'. ... * Makes chan_sip set the tag to the channel name. * Fixes received debug message sequence number. * Removed tx/rx debug message type since it was hard coded to 0. * Made udptl.c logged message header consistent if possible: "UDPTL (%s): ". * Removed unused rx_expected_seq_no from struct ast_udptl. (closes issue ASTERISK-18401) Reported by: Kevin P. Fleming Patches: jira_asterisk_18401_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Matthew Nicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@339625 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	2e320de4bf	Remove duplicated Maxforwards line in AMI output. ... (Closes issue ASTERISK-18637) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer.patch (License #6298) uploaded by Jacek Konieczny git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@339147 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	a0eb30ea43	Properly ignore AST_CONTROL_UPDATE_RTP_PEER in more places ... After the change in r336294, the new AST_CONTROL_UPDATE_RTP_PEER frame is sent when a re-invite happens. If we receive a re-invite from a device the waitstream_core was not aware of the new control frame and would drop the call. (closes issue ASTERISK-18610) Reported by: Kristijan_Vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@339086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4d9b980ab8	Fix segfault in analog_ss_thread() not checking ast_read() for NULL. ... NOTE: The problem was reported against v1.6.2. It is unlikely to ever happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely to be used. The version in sig_analog.c has largely replaced it. (closes issue ASTERISK-18648) Reported by: Stephan Bosch Patches: jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Stephan Bosch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338800 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c9546515e5	Fix formatting of AMI header for SIP show peer. ... ASTERISK-17486 exposed the problem for AMI parsers. (closes issue ASTERISK-18649) Reported by: Jacek Konieczny Patches: asterisk-sipshowpeer_response_end.patch (license #6298) patch uploaded by Jacek Konieczny git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	be71dfc76b	Update documentation for SIP_HEADER. ... The SIP_HEADER function only works on the the initial SIP INVITE. The documentation was updated in trunk, but not in 1.8 or 10, so I'm making them match. (Closes issue ASTERISK-18640) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	e7d6d7ee19	The rtptimeout setting is ignored on a per peer basis. ... Not only is the rtptimeout ignored in some cases but rtpkeepalive and rtpholdtimeout is affected. this commit also removes rtptimeout/rtpholdtimeout on text rtp. (closes issue ASTERISK-18559) Review: https://reviewboard.asterisk.org/r/1452 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8711d897d0	Make duplicate call ptr warning message more helpful. ... * Adds the value of the call ptr to the duplicate call ptr message to help trace why there is a duplicate call ptr. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	529ab3ad50	Add support levels to non-module sections of menuselect (cflags, utils, etc). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b535088ac6	Fix chan_dahd compiling with gcc 4.6 when PRI and SS7 not present. ... (closes issue ASTERISK-18357) Reported by: Matthew Nicholson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f8b799c0c1	Made ISDN not add numbering plan prefix strings to empty numbers. ... When the Caller-ID is restricted, the expected behavior is for the Caller-ID to be blank. In chan_dahdi, the national prefix is placed onto the Caller-ID number even if it is restricted (empty) causing the Caller-ID to be the national prefix rather than blank. This behavior was lost when sig_pri was extracted from chan_dahdi. * Made not add prefix strings to empty connected line, calling, and ANI number strings. (closes issue ASTERISK-18577) Reported by: Kris Shaw Patches: jira_asterisk_18577_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Kris Shaw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	c6dd0ef286	If IP address is used in chan_h323 host parameter of peer configuration. ... module tries to resolve IP address to IP address and fails. Simple fix to set family of socket this is a hangover from ipv6 changes. (closes issue ASTERISK-18237) (issue ASTERISK-17278) (issue ASTERISK-17500) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7361deae1b	Check if a channel was created before using the pointer in sig_ss7_new_ast_channel(). ... Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace. * Added some missing libss7 access lock protection. * Prevent cancelling the ss7_linkset() thread at inoportune times just like the pri_dchannel() thread. (issue ASTERISK-17955) Reported by: Ian M Sherman Patches: jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett (attached to related ASTERISK-17966) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@337007 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b48984e2fb	Fix deadlock from not releasing SS7 linkset lock. ... sig_ss7_hangup() failed to release the SS7 linkset lock if the call had the alreadyhungup flag set. * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the alreadyhungup flag is set. * Made ss7_start_call() not hold any locks while creating the channel for an incoming call to prevent deadlock. * Made ss7_grab() a void function, since it could never fail, to simplify calling code. * Made obtain the channel lock to do softhangup in some places. Patches: jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett JIRA AST-668 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336977 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	0628cce193	Don't interfere with T.38 reinvites ... This is an update to the fix for ASTERISK-18340 and ASTERISK-17725 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9eb7ccef76	Rework sig_pri_hangup() to be simpler and clearer. ... JIRA AST-675 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336569 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	535817fe71	Add diversion header to a 302 redirect response if we have diversion data ... (closes issue ASTERISK-18143) patch by oej git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	aa50191685	A long time ago in a galaxy far far away a IPv6 update was made, ... chan_h323 was not updated causeing all to flee to chan_ooh323. the brave Jedi [asterisk developers] pondered this miscarrige of justice and restored order to the force for the sake of closing out 2 old issues. (closes issue ASTERISK-17278) (closes issue ASTERISK-17500) Reported by: dread, sybasesql Tested by: irroot Reviewed by: IRC (russellb, kpfleming) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336499 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	7a2e489631	Add missing unlock at MWI message sending time ... (closes issue ASTERISK-18573) Patches: sip_mwi_lock.patch (license #5041) by Gregory Hinton Nietsky Thanks to irrot for the reminder, to Gregory for the patch! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	21714a05b6	Fix bad RTP media bridges in directmedia calls on peers separated by multiple Asterisk nodes. ... In a situation involving devices on separate Asterisk trunks, the remote RTP bridge would break when starting a call with directmedia. This patch queues a new type of control frame so that our RTP bridge loop can properly detect when these situations occur and check to see if peers need to be updated in order to send their media to the proper location. (Closes issue ASTERISK-18340) Reported by: Thomas Arimont (Closes issue ASTERISK-17725) Reported by: kwk Tested by: twilson, jrose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	bbc088b9fc	The round robin routing routine in chan_misdn.c is broken. ... it rotates between ports but never checks the channels in the ports. i have extensivly tested it and verified it works on 1 upto 4 ports. before the patch only 1 out of each port was used now all are used as expected. (closes issue ASTERISK-18413) Reported by: irroot Tested by: irroot Reviewed by: irroot Review: https://reviewboard.asterisk.org/r/1410/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@336166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	46e2968917	lock the channel before calling ast_bridged_channel() to prevent a seg fault. ... AMI agents list called on shutdown causes a segfault, introducing proper locking will prevent this. (closes issue ASTERISK-18092) Reported by: agustina Patches: chan_agent.patch (License #5041) patch uploaded by irroot git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b695a91265	Fixed cut-n-paste regression using the wrong variable. ... Fixes the missing DAHDI channels when using the newer chan_dahdi.conf sections for channel configuration. (closes issue ASTERISK-18496) Reported by: Sean Darcy Patches: jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Sean Darcy, rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	b1b865d7b2	Prevent IAX2 from getting IPv6 addresses via DNS ... IAX2 does not support IPv6 and getting such addresses from DNS can cause error messages on the remote end involving bad IPv4 address casts in the presence of IPv6/IPv4 tunnels. This patch ensures that IAX2 will not encounter IPv6 addresses via DNS queries. (closes issue ASTERISK-18090) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Olle Johansson	c0ab1f3281	Lock the peer->mvipvt to avoid crashes with SIP history enabled ... After the launch of 1.6 event-based MWI we have two threads handling the peer->mwipvt, which cause issues with SIP history additions in combination with the max limit for number of history entries. Review: https://reviewboard.asterisk.org/r/1373/ (closes issue ASTERISK-18288) Thanks to irrot for peer review. Work with this bug funded by IPvision AS git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Stefan Schmidt	22b30eb82c	build_peer doesnt unlink a peer object from peers_by_ip container which leads to a wrong refcounter value. ... adding an ao2_unlink from the peers_by_ip container fix it. Review: https://reviewboard.asterisk.org/r/1428/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7dc49195d8	Updated SIP 484 handling; added Incomplete control frame ... When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	f105f3e579	Cleanup chan_iax2.c log messages ... Review: https://code.asterisk.org/code/cru/CR-AST-11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334843 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago