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Author	SHA1	Message	Date
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	2ae86be1f2	Fix AST_LIST_INSERT_BEFORE_CURRENT() updating the wrong variable. ... AST_LIST_INSERT_BEFORE_CURRENT() could not be used twice in an iteration or before AST_LIST_REMOVE_CURRENT() without corrupting the list. AST_LIST_INSERT_BEFORE_CURRENT() could also corrupt the list if AST_LIST_INSERT_BEFORE_CURRENT() or AST_LIST_REMOVE_CURRENT() is used on the next iteration. * Fixed cut and paste error using the wrong variable in AST_LIST_INSERT_BEFORE_CURRENT(). * Added linked list unit tests for AST_LIST_INSERT_BEFORE_CURRENT(), AST_LIST_APPEND_LIST(), and AST_LIST_INSERT_LIST_AFTER(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	16c9d2231b	Fix sequence number overflow over 16 bits causing codec change in RTP packets. ... Sequence number was handled as an unsigned integer (usually 32 bits I think, more depending on the architecture) and was put into the rtp packet which is basically just a bunch of bits using an or operation. Sequence number only has 16 bits allocated to it in an RTP packet anyway, so it would add to the next field which just happened to be the codec. This makes sure the sequence number is set to be a 16 bit integer regardless of architecture (hopefully) and also makes it so the incrementing of the sequence number does bitwise or at the peak of a 16 bit number so that the value will be set back to 0 when going beyond 65535 anyway. (closes issue ASTERISK-18291) Reported by: Will Schick Review: https://reviewboard.asterisk.org/r/1542/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	dffe914c06	Cleanup reference leaks in res_jabber ... res_jabber.c had a number of places where astobjs would be referenced and have their reference counts bumped without having a dereference made before the object lost scope. This patch adds a number of ASTOBJ_UNREFs to resolve that. Review: https://reviewboard.asterisk.org/r/1478/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d109a3973b	Check fopen return value for ao2 reference debug output. ... Reported by: wdoekes Patched by: wdoekes Review: https://reviewboard.asterisk.org/r/1539/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342487 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
Terry Wilson	e9dc0ae56d	Use int for storing ao2_container_count instad of size_t ... AST-676 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	33f73e2ae2	Simplify queue membercount code ... Despite an ominous sounding comment stating that membercount was for "logged in" members only and thus we couldn't use ao2_container_count(), I could not find a single place in the code where that seemed to be accurate. The only time we decremented membercount was when we were marking something dead or actually removing it. The only places we incremented it were either after ao2_link(), or trying to correct for having set it to 0 during a reload. In every case where we were correcting the value, it seemed that we were trying to make the count actually match what ao2_container_count() would return. The only place I could find where we made a determination about something being "logged in" or not, we didn't trust the membercount, but instead looked at devicestate, paused, etc. This patch removes membercount, replaces its use with ao2_container_count, and manually adds the results of ao2_container_count to a "membercount" field for ast_data queue query results. This patch also would fix AST-676, but as it is slightly riskier than the previously committed fix, the two commits have been made separately. Reivew: https://reviewboard.asterisk.org/r/1541/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f927ef5571	Properly update membercount for reloaded members ... Since q->membercount is set to 0 before reloading, it is important to increment it again for reloaded members as well as added. (closes issue AST-676) Review: https://reviewboard.asterisk.org/r/1541/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	5ee32dfa2c	Fix compilation on Snow Leopard/FreeBSD for pbx_spool.c ... One of the changes in the recent spool handling of hardlinks patch was just outside a HAVE_INOTIFY block and caused compilation to fail in some build environments. This has been corrected. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	40891b278f	Fix spool handling to allow call files to be hardlinked into place ... This fixes the inotify code to handle call files being hardlinked into the spool directory. The smsq utility does this, instead of rename(), to ensure that it cannot accidentally overwrite an existing spool file. A rename() might do that, but link() will definitely not. The inotify code had broken this, because it would wait for an IN_CLOSE_WRITE event on the file... which was never forthcoming, since it was never opened. Now we look for IN_OPEN events following the IN_CREATE event, and only wait for an IN_CLOSE_WRITE if the file was actually opened. Patch-by: dwmw2 (closes issue ASTERISK-18331) Review: https://reviewboard.asterisk.org/r/1391/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	a98dd1933b	Return NULL when no results returned for realtime_multientry ... It was not documented what the return value should be when no entries were returned with the multientry realtime callback. This change forces consistent behavior even if the backends return an empty ast_config. Review: https://reviewboard.asterisk.org/r/1521/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342223 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
Gregory Nietsky	4037f1366d	Revert Janitor patch 341906 For now ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	4f690341cf	Whitespace Fixups / Add Braces ... This janitorial patch is related to work on RB1538 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	a16caab0e0	only process args that exist ... ASTERISK-18395 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Nicholson	c0d3d4b0e4	don't limit the length of app and function arguments ... ASTERISK-18395 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	dbacd97e17	Fix AGI exec Park to honor the Park application parameters. ... The fix for ASTERISK-12715 and ASTERISK-12685 added a check for the Park application because the channel needed to be masqueraded to prevent a crash. Since the Park application now always masquerades the channel into the parking lot, the special check is no longer needed. The fix also resulted in AGI exec Park attempting to double park the call and not honor the Park application parameters. * Removed no longer necessary call to ast_masq_park_call() by AGI exec for the Park application. (Reverts -r146923) * Fix Park application to only return 0 or -1. The AGI exec Park was causing broken pipe error messages because the Park application returned 1 on successful park. (closes issue ASTERISK-18737) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	f9addb13d9	Fixed typo from previous commit ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341704 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	cc70599f21	Updated documentation for the optional CID parameter with CALLERID ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341664 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	c86eaf3028	Clean up ast_check_digits ... The code was originally copied from the is_int() function in the AEL code. wdoekes pointed out that the function should take a const char* and that their was an unneeded variable. This is now fixed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341529 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
Alexandr Anikin	0c360ed6be	fix issue on channel numbering (calls could have same channel number ... on heavy loaded system) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341312 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
Paul Belanger	fb6e8a5575	Fix previous commit ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	902b38d21d	Voicemail compiler flags are 'core' support ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341108 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
Tzafrir Cohen	499262c2b3	Remove an unused include of md5.h ... Unused include of asterisk/md5.h in pbx_realtime.c . A commit needed to test the commit message. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341074 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	25bc68ac1e	Change the internal name of the menuselect options that are used to control ... whether modules are embedded or not; using just the bare category name led to accidentally enabling these options when users used the wrong "--enable" operation on the menuselect command line. Now the internal option names are prefixed with "EMBED_", so they won't be the same as the name of the category containing the modules they control the embedding of. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341022 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
Terry Wilson	927336fe2f	Avoid unnecessary WARNING message ... Add AST_CONTROL_UPDATE_RTP_PEER frame to be ignored here to avoid displaying a WARNING message. (closes issue ASTERISK-18610) Patch by: Kristijan_Vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340878 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	88bf8d3316	Fixes some support level info so that it can be read by menuselect. ... (issue ASTERISK-18268) Review: https://reviewboard.asterisk.org/r/1525/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	85c808bfc4	Fix DTMF blind transfer continuing to execute dialplan after transfer. ... Party A calls Party B. Party A DTMF blind transfers Party B to Party C. Party A channel continues to execute dialplan. * Fixed the return value of builtin_blindtransfer() to return the correct value after a transfer so the dialplan will not keep executing. * Removed unnecessary connected line update that did not really do anything. * Made access to GOTO_ON_BLINDXFR thread safe in check_goto_on_transfer(). * Fixed leak of xferchan for failure cases in check_goto_on_transfer(). * Updated debug messages in builtin_blindtransfer() and check_goto_on_transfer(). (closes issue ASTERISK-18275) Reported by: rmudgett Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340809 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
Terry Wilson	631c6a9a74	Don't skip the query field on a realtime multi query ... There is no documented reason to not add the query field to the varlist returned by a realtime multi query, despite the config category being set to its value. Of course, there is no documentation that the category should be set to the value either. There is lots of no documentation when it comes to realtime. But, other engines do not skip this field so I am forcing this backend to follow the convention, because not doing so is very silly. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340662 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
Richard Mudgett	4051609b5c	Update MeetMe p and X option documentation when interacting with the s option. ... ASTERISK-12175 changed the p and X options to not interfere with the s option when they are used together. It makes more sense for the s option to have priority for the DTMF '*' key since it cannot change its activation code. Otherwise, you could not use option s with the p or X options. JIRA AST-671 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340470 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
Richard Mudgett	236637104d	Convert registered AMI actions to ao2 objects. ... * Fixed race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. * Fixed potential memory leak if an AMI action failed to get registered because is already was registered. Part of the ao2 conversion. * Fixed AMI ListCommands action not walking the actions list with a lock held. * Fix usage of ast_strdupa() and alloca() in loops. Excess stack usage. * Fix AMI Originate action Variable header requiring a space after the header colon. Reported by Yaroslav Panych on the asterisk-dev list. * Increased the number of listed variables allowed per AMI Originate action Variable header to 64. * Fixed AMI GetConfigJSON action output format. * Fixed usage of res contents outside of scope in append_channel_vars(). * Fixed inconsistency of config file channelvars option. The values no longer accumulate with every channelvars option in the config file. Only the last value is kept to be consistent with the CLI "manager show settings" command. (closes issue ASTERISK-18479) Reported by: Jaco Kroon git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	9a426fab4d	Update SHA1 code to RFC 6234 ... RFC 6234 is an update to RFC 3174 from which the code was originally taken. It has a slightly better code, and a better phrased license (simple 3-clause BSD). * main/sha1.c is sha1.c from RFC 6234 with formatting changes only. * include/asterisk/sha1.h merges sha.h and sha-private.h from RFC 6234. * Removed unused include of asterisk/sha1.h from main/channels.c Review: https://reviewboard.asterisk.org/r/1503/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340263 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