mirror of https://github.com/asterisk/asterisk
master
20
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${ noResults }
1712 Commits (a7733c579b25c8d5c30b77e76e21e5dc46a5420a)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
7eb528fbf1 |
Fix checking bounds of array index after using it; improper sizeof
This patch fixes two problems pointed out by a static analysis tool. * In chan_dahdi, when an event is handled the index of the sub channel is first obtained. In very off nominal cases, the method that determines the index can return a negative value. In the event handling code, whether or not the index returned is valid was being checked after that value was used to index into an array. This patch makes it so the value is checked before any indexing is done. * In res_calendar_ews, sizeof was being passed a pointer instead of the struct to determine the amount of memory to allocate. (issue ASTERISK-19651) Reported by: Matt Jordan (closes issue ASTERISK-19671) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366740 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
|
|
a94fcae21b |
Resolve FORWARD_NULL static analysis warnings
This resolves core findings from ASTERISK-19650 numbers 0-2, 6, 7, 9-11, 14-20, 22-24, 28, 30-32, 34-36, 42-56, 82-84, 87, 89-90, 93-102, 104, 105, 109-111, and 115. Finding numbers 26, 33, and 29 were already resolved. Those skipped were either extended/deprecated or in areas of code that shouldn't be disturbed. (Closes issue ASTERISK-19650) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ae528efea3 |
Coverity Report: Fix issues for error type CHECKED_RETURN for core
(issue ASTERISK-19658) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1905/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@366094 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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eafb430562 |
Fix findings 0-3, 5, and 8 for Coverity MISSING_BREAK errors.
(Issue ASTERISK-19662) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c6d9524482 |
Fix core FINDING 2, FINDING 3, and FINDING 4 from Coverity's CONSTANT_EXPRESSION_RESULT report.
These three all are in RTP code that attempts to print the number of sequence number cycles in an RTCP RR report. The code was masking out the upper 16 bits and then shifting the number right by 16 bits. This led to an all zero result in all cases. The fix is to do the shift without the bit masking. (issue ASTERISK-19649) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@365298 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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70bde6ffa7 |
Fix places in resources where a negative return value could impact execution
This patch addresses a number of modules in resources that did not handle the negative return value from function calls adequately. This includes: * res_agi.c: if the result of the read function is a negative number, indicating some failure, the result would instead be treated as the number of bytes read. This patch now treats negative results in the same manner as an end of file condition, with the exception that it also logs the error code indicated by the return. * res_musiconhold.c: if spawn_mp3 fails to assign a file descriptor to srcfd, and instead assigns a negative value, that file descriptor could later be passed to functions that require a valid file descriptor. If spawn_mp3 fails, we now immediately retry instead of continuing in the logic. * res_rtp_asterisk.c: if no codec can be matched between two RTP instances in a peer to peer bridge, we immediately return instead of attempting to use the codec payload type as an index to determine the appropriate negotiated codec. (issue ASTERISK-19655) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1863/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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bcd63be3cd |
Make use of va_args more appropriate to form in various res_config modules plus utils.
A number of va_copy operations weren't matched with a corresponding va_end in res_config_odbc. Also, there was a potential for va_end to be invoked twice on the same va_arg in utils, which would mean invoking va_end on an undefined variable... which is bad. va_end is removed from various functions in config_pgsql and config_curl since they aren't making their own copy. The invokers of those functions are responsible for calling va_end on them. (issue ASTERISK-19451) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/1848/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@362354 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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1d3ba1f2a7 |
Fix memory leak in res_calendar_ews when event email address node is empty
If the XML calendar data returned by a Microsoft Exchange Web Service
specifies an XML Event E-Mail Address ("EmailAddress"), and no e-mail address
is provided, a condition existed where an ast_calendar_attendee struct would
be allocated but not appended to the list of attendees. Because of that,
the memory associated with the attendee would never be freed. This patch
frees the memory if no e-mail address is provided.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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4148e51555 |
Add missing newlines to CLI logging
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@361471 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2b2c5f27fc |
Rebuild parsers.
This is needed to include the last fix to main/ast_expr2.y. The changes look much bigger as this regeneration of the code was done with newer versions of flex and bison. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7bff9be851 |
Fix incorrect usage of sizeof() in res_crypto.
In this case, just remove the memset(). There was a redundant memset that is done correctly just 2 lines later. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359110 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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30d3e59447 |
Fix broken usage of sizeof() in res_adsi.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ac53c0d94e |
Dump cache of published events when a node joins the cluster.
Also use a more reliable method for stopping the poll() thread. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359053 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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2477215dd4 |
Enable macros in 1.8 to find the next highest "h" extension in a context, like in 1.4.
This change restores functionality that was present in 1.4, when AEL macros were implemented with the Macro dialplan application. Macros are fraught with functionality issues, because they consume a large portion of the underlying application stack. This limits the ability of AEL users to call many layers of subroutines, an issue which Gosub does not have (originally tested to 100,000 levels deep). Therefore, starting in 1.6.0, AEL macros were implemented with Gosub. However, there were some implicit behaviors of Macro, which were not replicated at the same time as with the transition to Gosub, one of which is documented in the related issue. In particular, the "h" extension is designed to execute not in the Macro context, but in the topmost calling context. Due to legacy issues with a misapplied bugfix many years ago, when a macro exited in 1.4, it looks in all calling contexts, bubbling up from the deepest level until it finds an "h" extension. Since AEL hides the complexity of the underlying dialplan logic from the AEL programmer, it's reasonable to assume that this behavior should not change in the transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we break working AEL configurations in the transition to Asterisk 1.8 LTS. This fix is the result, which implements a search for the "h" extension in all calling Gosub contexts. Fixes ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff (License #5003) by Tilghman Lesher (with slight modifications for 1.8) Tested by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1776/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358810 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c7d587dd49 |
Remove possible segfaults from res_odbc by adding locks around usage of odbc handle
(closes issue ASTERISK-19011) Reported by: Walter Doekes Patches: issueA19011_combine_read_and_write_locks_WORK_IN_PROGRESS.patch uploaded by Walter Doekes (license 5674) review: https://reviewboard.asterisk.org/r/1719/ review: https://reviewboard.asterisk.org/r/1622/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ed81d0e585 |
Remove srtp_shutdown from res_srtp
The patch for ASTERISK-19253 included properly shutting down the libsrtp library in the case of module unload. Unfortunately, not all distributions have the srtp_shutdown call. As such, this patch removes calling srtp_shutdown. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356650 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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032962f1a2 |
Allow SRTP policies to be reloaded
Currently, when using res_srtp, once the SRTP policy has been added to the current session the policy is locked into place. Any attempt to replace an existing policy, which would be needed if the remote endpoint negotiated a new cryptographic key, is instead rejected in res_srtp. This happens in particular in transfer scenarios, where the endpoint that Asterisk is communicating with changes but uses the same RTP session. This patch modifies res_srtp to allow remote and local policies to be reloaded in the underlying SRTP library. From the perspective of users of the SRTP API, the only change is that the adding of remote and local policies are now added in a single method call, whereas they previously were added separately. This was changed to account for the differences in handling remote and local policies in libsrtp. Review: https://reviewboard.asterisk.org/r/1741/ (closes issue ASTERISK-19253) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283) (with some small modifications for this check-in) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356604 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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9a3c569772 |
Track module use count for res_calendar
If the res_calendar module was followed immediately by one of the calendar tech modules and "core stop gracefully" was run, Asterisk would crash. This patch adds use count tracking for res_calendar so that it is unloaded after the tech modules when shutting down gracefully. It is now not possible to unload all the of the calendar modules via "module unload res_calednar.so", but it is still possible to unload them all via "module unload -h res_calendar.so". Review: https://reviewboard.asterisk.org/r/1752/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356291 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0b894ef73a |
Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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695e1ecbec |
Fix AMI Monitor action without File header converting channel name into filename.
* Fix potential Solaris crash if Monitor application has a urlbase and no fname_base option. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7d84a0ed69 |
Fix reconnecting to pgsql database after connection loss.
There can only be one database connection in res_config_pgsql just like res_config_sqlite. If the connection is lost, the connection may not get reestablished to the same database if the res_pgsql.conf and extconfig.conf files are inconsistent. * Made only use the configured database from res_pgsql.conf. * Fixed potential buffer overwrite of last[] in config_pgsql(). (closes issue ASTERISK-16982) Reported by: german aracil boned Review: https://reviewboard.asterisk.org/r/1731/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354953 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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aeadbab220 |
Adding reload support to res_fax.so
(closes issue ASTERISK-16712) reported by Frank DiGennaro Review: https://reviewboard.asterisk.org/r/1713 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@354545 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0120bde7d8 |
Allow res_calendar to be unloaded
The calendaring tech modules depend on res_calendar and initially res_calendar just bumped the use count so that it couldn't be unloaded. res_calendar can potentially create many threads and I've seen issues where the Asterisk shutdown has failed where it looked like these threads could be the culprit. This patch adds unload support for res_calendar. Unloading res_calendar will also unload the dependant tech modules as well. (closes issue ASTERISK-16744) Review: https://reviewboard.asterisk.org/r/1657/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@353502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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cda638dcfc |
Make failed PauseMonitor and UnpauseMonitor with no valid channel not close AMI session.
I also went ahead and took a little time to make sure that the manager value AMI_SUCCESS was used instead of just return 0 being thrown around everywhere since that's how we handle this stuff these days. (closes issue ASTERISK-19249) Reporter: Jamuel Starkey Patches: res_monitor.c-ASTERISK-19249.diff uploaded by Jamuel Starkey (license 5766) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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a55030f4fa |
Audit of ao2_iterator_init() usage for v1.8.
Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352955 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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11105d0cd3 |
Correctly apply FAXOPT settings (V17, V27, V29) before starting spandsp layer
While the FAXOPT function could be used to set the modem capabilities, the input to that function was not being applied correctly to the spandsp layer. This patch applies the current model capabilities before starting the spandsp layer. (closes issue: ASTERISK-16409) Reported by: Kristijan Vrban Tested by: Matt Jordan, Matthew Nicholson Patches: spandsp-modems-1.8.diff uploaded by mnicholson (license 5081) spandsp-modems-10.diff uploaded by mnicholson (license 5081) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@352144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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46fce6837a |
Correct output of RTCP jitter statistics in SR and RR reports
Change the RTCP RR and SR generation code to convert Asterisk's internal jitter statistics to be represented in RTP timestamp units based on the rate of the codec in use instead of in seconds. (closes issue ASTERISK-14530) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351611 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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37a8ff4dc8 |
Eliminate odd initialization of probation variable.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351306 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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6aa2cd51f9 |
Adds pjmedia probation concepts to res_rtp_asterisk's learning mode.
In order to better handle RTP sources with strictrtp enabled (which is now default in 10) using the learning mode to figure out new sources when they change is handled by checking for a number of consecutive (by sequence number) packets received to an rtp struct based on a new configurable value called 'probation'. Also, during learning mode instead of liberally accepting all packets received, we now reject packets until a clear source has been determined. Review: https://reviewboard.asterisk.org/r/1663/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@351287 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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0ee313f076 |
Fix timing source dependency issues with MOH
Prior to this patch, res_musiconhold existed at the same module priority level as the timing sources that it depends on. This would cause a problem when music on hold was reloaded, as the timing source could be changed after res_musiconhold was processed. This patch adds a new module priority level, AST_MODPRI_TIMING, that the various timing modules are now loaded at. This now occurs before loading other resource modules, such that the timing source is guaranteed to be set prior to resolving the timing source dependencies. (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H, Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont Patches: asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-1.8.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3_branch-10.diff uploaded by elguero (License #5026) asterisk-17474-dahdi_timing-infinite-wait-fix_v3.diff uploaded by elguero (License #5026) Review: https://reviewboard.asterisk.org/r/1578/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@349194 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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74da7648bb |
Fix crash during CDR update.
The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348362 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5921a96450 |
Don't clear LOCALSTATIONID before sending or receiving. The user may set that
variable. ASTERISK-18921 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@348212 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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607398d450 |
Add a separate buffer for SRTCP packets
The function ast_srtp_protect used a common buffer for both SRTP and SRTCP packets. Since this function can be called from multiple threads for the same SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the packets to become corrupted as the buffer was used by both threads simultaneously. This patch adds a separate buffer for SRTCP packets to avoid the problem. (closes issue ASTERISK-18889, Reported/patch by Daniel Collins) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347995 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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665581beff |
Fix chan_jingle/gtalk load regression introduced in r346087
Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy for usage outside res_jabber. Testing of these changes focused on res_jabber itself, so this problem was missed. Reported-by: Michael Spiceland git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c70441c168 |
Re-resolve the STUN address if a STUN poll fails for res_stun_monitor.
The STUN socket must remain open between polls or the external address seen by the STUN server is likely to change. However, if the STUN request poll fails then the STUN server address needs to be re-resolved and the STUN socket needs to be closed and reopened. * Re-resolve the STUN server address and create a new socket if the STUN request poll fails. * Fix ast_stun_request() return value consistency. * Fix ast_stun_request() to check the received packet for expected message type and transaction ID. * Fix ast_stun_request() to read packets until timeout or an associated response packet is found. The stun_purge_socket() hack is no longer required. * Reduce ast_stun_request() error messages to debug output. * No longer pass in the destination address to ast_stun_request() if the socket is already bound or connected to the destination. (closes issue ASTERISK-18327) Reported by: Wolfram Joost Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1595/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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7b3a04cb6f |
Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified.
(closes issue ASTERISK-18693) Reported by: Davide Dal Fra Review: https://reviewboard.asterisk.org/r/1600/ Reviewed by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c4b7983866 |
Fix res_jabber resource leaks
This should fix almost all resource leaks in res_jabber that involve ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where ast_aji_get_client would sometimes bump an object's refcount and sometimes not. Review: https://reviewboard.asterisk.org/r/1553 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346086 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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ac656bc6fe |
Resume playing existing hold music for cached realtime MOH
As a result of the fix for ASTERISK-18039, realtime caching MOH no longer properly resumes playing back a file between different holds in the same call. This is because scanning for new files causes the existing file array to be emptied and we were just comparing that the saved pointer to the filename matched the pointer to the filename in a particular position in the array. An easy fix is to save the filename instead of a pointer to it and then do a strcmp instead of comparing the addresses. (closes issue ASTERISK-18912) Review: https://reviewboard.asterisk.org/r/1596/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@346030 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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37611b4ecc |
Make FastAGI HANGUP show up in AGI debug output.
* Change from using send() to ast_agi_send() so the HANGUP shows up in the
AGI debug output.
(closes issue ASTERISK-18723)
Reported by: James Van Vleet
Patches:
jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@345431 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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14 years ago |
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6c27911d5b |
Don't forget to rescan MOH files for cached realtime classes
Realtime MOH class caching was implemented because without it, you would build a completely new MOH class and would start the music over at the beginning each time hold was pressed in a conversation. Unfortunately, this broke re-scanning for file changes for realtime MOH classes. This patch corrects that issue. (closes issue ASTERISK-18039) Review: https://reviewboard.asterisk.org/r/1579/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344899 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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b4ad988a5a |
only attempt to do stun handling on ipv4 or ipv4 mapped to ipv6 addresses
Patch by: jkonieczny (modified) ASTERISK-18490 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@344330 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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d5238279e7 |
Fix sqlite config driver segfault and broken queries
The sqlite realtime handler assumed you had a static config configured as well. The realtime multientry handler assumed that you weren't using dynamic realtime. (closes issue ASTERISK-18354) (closes issue ASTERISK-18355) Review: https://reviewboard.asterisk.org/r/1561 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@343375 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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68b523463d |
Cleanup references to sipusers and sipfriends dynamic realtime families
Somewhere between 1.4 and 1.8 the sipusers family has become completely unused. Before that, the sipfriends family had been obsoleted in favor of separate sipusers and sippeers families. Apparently, they have been merged back again into a single family which is now called "sippeers". Reviewed by: irroot, oej, pabelanger Review: https://reviewboard.asterisk.org/r/1523 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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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 |
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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 |
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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 |
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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 |
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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 |
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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 |
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de9e8e501e |
Load the proper XML documentation when multiple modules document the same application.
This patch adds an optional "module" attribute to the XML documentation spec that allows the documentation processor to match apps with identical names from different modules to their documentation. This patch also fixes a number of bugs with the documentation processor and should make it a little more efficient. Support for multiple languages has also been properly implemented. ASTERISK-18130 Review: https://reviewboard.asterisk.org/r/1485/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340108 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |