7051 Commits (d63e58203a796bb59e20b51cb705254100621c9a)

Author	SHA1	Message	Date
Richard Mudgett	d63e58203a	Fixed reference to incorrect variable if unknown host configured crash. ... * Fixed a LOG_ERROR message referencing the config variable list v that had previously been processed and became NULL. * Added error return value set that was missing in an ast_append_ha() error return path. (closes issue ASTERISK-18743) Reported by: Michele Patches: issueA18743-fix_dynamic_exclude_static_bad_host_log.patch (license #5674) patch uploaded by Walter Doekes Tested by: Michele git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	59f002fd67	respect case changes in peer names on sip reload ... ASTERISK-18669 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	71769a3245	Fix __sip_subscribe_mwi_do() incorectly changing dialogs hash key callid. ... Changing an object value used as a container key requires removing the object from the container and reinserting it. * Created change_callid_pvt() to call instead of build_callid_pvt(). The change_callid_pvt() will correctly change the dialog callid so the ao2 conainter can explicitly unlink it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	dc3ca68670	Prevent BLF subscriptions from causing deadlocks ... Fix a locking inversion in sip_send_mwi_to_peer that was causing deadlocks. This function now requires that both the peer and associated pvt be unlocked before it is called for cases where peer and peer->mwipvt form a circular reference. (closes issue ASTERISK-18663) Review: https://reviewboard.asterisk.org/r/1563/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343621 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3864c4083c	Fix deadlock if peer is destroyed while sending MWI notice. ... A dialog cannot be destroyed by the ao2_callback dialog_needdestroy because of a deadlock between the dialogs container lock and the RWLOCK of the events subscription list. * Create dialogs_to_destroy container to hold dialogs that will be destroyed. * Ensure that the event subscription callback will never happen with an invalid peer pointer by making the event callback removal the first thing in the peer destructor callback. (closes issue ASTERISK-18747) Reported by: Gregory Hinton Nietsky Review: https://reviewboard.asterisk.org/r/1564/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	702726c917	Make room for the fax detect flags ... The original REGISTERTRYING flag, in addition to being impossible to check, also encroached on the space for the flag above it. This patch moves the flags that were below REGISTERTRYING back to where they were as though we had just removed the REGISTERTRYING option. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5844133aba	Remove registertrying option in chan_sip ... This option is not only useless, but has been broken since inception since the flag was never copied from the peer where it is set to the pvt where it was checked. RFC 3261 specificially states that you should not send a provisional response to a non-INVITE request, and if we did fix the code so that it worked, it would cause the same kind of user enumeration vulnerability that we've discussed with the nat= setting. This patch removes registertrying option and any code that would have sent a 100 response to a register. Review: https://reviewboard.asterisk.org/r/1562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	fb50c1db4b	Fix improper warning introduced by r342927 and more tweaks ... Changeset r342927 introduced a warning which was only supposed to be emitted when a found realtime peer had an empty (or no) name. It turned out that there were some inconsistencies left. Now found peers with an empty name are explicitly ignored like before r342927 but better. Reviewed by: Stefan Schmidts, Terry Wilson Review: https://reviewboard.asterisk.org/r/1560 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343181 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	ab2dacb555	Several fixes to the chan_sip dynamic realtime peer/user lookup ... There were several problems with the dynamic realtime peer/user lookup code. The lookup logic had become rather hard to read due to lots of incremental changes to the realtime_peer function. And, during the addition of the sipregs functionality, several possibilities for memory leaks had been introduced. The insecure=port matching has always been broken for anyone using the sipregs family. And, related, the broken implementation forced those using sipregs to *still* have an ipaddr column on their sippeers table. Thanks Terry Wilson for comprehensive testing and finding and fixing unexpected behaviour from the multientry realtime call which caused the realtime_peer to have a completely unused code path. This changeset fixes the leaks, the lookup inconsistenties and that you won't need an ipaddr column on your sippeers table anymore (when you're using sipregs). Beware that when you're using sipregs, peers with insecure=port will now start matching! (closes issue ASTERISK-17792) (closes issue ASTERISK-18356) Reported by: marcelloceschia, Walter Doekes Reviewed by: Terry Wilson Review: https://reviewboard.asterisk.org/r/1395 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	8765a80355	Fixed invalid memory access when adding extension to pattern match tree ... When an extension is removed from a context, its entry in the pattern match tree is not deleted. Instead, the extension is marked as deleted. When an extension is removed and re-added, if that extension is also a prefix of another extension, several log messages would report an error and did not check whether or not the extension was deleted before accessing the memory. Additionally, if the extension was already in the tree but previously deleted, and the pattern was at the end of a match, the findonly flag was not honored and the extension would be erroneously undeleted. Additionaly, it was discovered that an IAX2 peer could be unregistered via the CLI, while at the same time it could be scheduled for unregistration by Asterisk. The unregistration method now checks to see if the peer was already unregistered before continuing with an unregistration. (closes issue ASTERISK-18135) Reported by: Jaco Kroon, Henry Fernandes, Kristijan Vrban Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1526 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
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	15 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	15 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	15 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	15 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	15 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	15 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	15 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	15 years ago