mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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certified/1.8.11-cert1
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${ noResults }
23524 Commits (c2e28fae9cce3abb28bc0f8b09b00a4f046f83d5)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Jonathan Rose
|
c2e28fae9c |
Put clicompat-r2.patch back in
We've figured out how to resolve the problems this was causing in 12/trunk,
so this can go back in now.
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
clicompat-r2.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 401914 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Jonathan Rose
|
c10d4cf11c |
revert clicompat-r2.patch from r401704
Patch caused the following build errors against testsuite https://bamboo.asterisk.org/bamboo/browse/AST-ATRUNKBUILD4-244 (issue ASTERISK-22467) Reported by: Corey Farrell ........ Merged revisions 401895 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401896 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Kevin Harwell
|
b0288cc8ad |
chan_sip: Allow a sip peer to accept both AVP and AVPF calls
Adapts the behaviour of avpf to only impact the format of outgoing calls. For
inbound calls, both AVP and AVPF calls will be accepted regardless of the value
of avpf in the configuration.
(closes issue ASTERISK-22005)
Reported by: Torrey Searle
Patches:
optional_avpf_trunk.patch uploaded by tsearle (license 5334)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Kevin Harwell
|
cb6492b7e2 |
Logging: Logging types ignored after specifying a verbose level
If one specified a verbose level within a logging facility in logger.conf then any component after it was ignored. Fixed so all values are correctly read. (closes issue ASTERISK-22456) Reported by: Kevin Harwell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401833 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
a941dc3159 |
utils: Fix memory leaks and missed unregistration of CLI commands on shutdown
Final set of patches in a series of memory leak/cleanup patches by Corey Farrell
(closes issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
main-utils-1.8.patch uploaded by coreyfarrell (license 5909)
main-utils-11.patch uploaded by coreyfarrell (license 5909)
main-utils-12up.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 401829 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Jonathan Rose
|
b59d4497fe |
test_linkedlists: Fix memory leak
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
test_linkedlists-1.8.patch uploaded by coreyfarrell (license 5909)
test_linkedlists-11up.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 401790 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401791 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Jonathan Rose
|
19457016bd |
jitterbuf: Fix memory leak on jitter buffer reset
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
jitterbuf-jb_reset-leak-1.8.patch
jitterbuf-jb_reset-leak-11up.patch
........
Merged revisions 401786 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Jonathan Rose
|
d4da5d79bf |
astobj2: Unregister debug CLI commands at exit
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
astobj2-clean-debug-cli-1.8-11.patch uploaded by coreyfarrell (license 5909)
astobj2-clean-debug-cli-12up.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 401781 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Jonathan Rose
|
9dae9a5644 |
app_voicemail: Memory Leaks against tests
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
app_voicemail-1.8.patch uploaded by coreyfarrell (license 5909)
app_voicemail-11up.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 401743 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Jonathan Rose
|
037dee9eaa |
memory leaks: Memory leak cleanup patch by Corey Farrell (second set)
Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was
replaced with one of my own.
(issue ASTERISK-22467)
Reported by: Corey Farrell
Patches:
chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909)
clicompat-r2.patch uploaded by coreyfarrell (license 5909)
codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909)
data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909)
main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 401704 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Jonathan Rose
|
bab6b3610a |
memory leaks: Memory leak cleanup patch by Corey Farrell (first set)
(issue ASTERSIK-22467)
Reported by: Corey Farrell
Patches:
chan_sip-parse_contact_header_test-free-contacts.patch uploaded by coreyfarrell (license 5909)
cli-filename-completion-leak.patch uploaded by coreyfarrell (license 5909)
func_math.patch uploaded by corefarrell (license 5909)
main-test-cleanup.patch uploaded by coreyfarrell (license 5909)
test_dlinklists.patch uploaded by coreyfarrell (license 5909)
........
Merged revisions 401660 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Jonathan Rose
|
4241f7b6ea |
res_rtp_asterisk: Address jittery DTMF events in RTP streams
(closes issue ASTERISK-21170)
Reported by: NITESH BANSAL
Patches:
dtmf-timestamp.patch uploaded by NITESH BANSAL (license 6418)
Review: https://reviewboard.asterisk.org/r/2938/
........
Merged revisions 401619 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
Richard Mudgett
|
4e5291c1db |
cdr_adaptive_odbc: Also apply a filter when the CDR value is empty.
Extra CDR records are written if a filtered CDR value is empty because the filter is not checked. (closes issue ASTERISK-22272) Reported by: Jordi Llull Chavarria ........ Merged revisions 401577 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401579 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Kinsey Moore
|
c932415383 |
chan_mgcp: Properly handle malformed media lines
This corrects a situation in which a media line was not parsed properly
and resulted in a crash.
(closes issue ASTERISK-21190)
Reported by: adomjan
Patches:
chan_mgcp.c-sscnaf_fix uploaded by adomjan (License 5448)
........
Merged revisions 401537 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
Joshua Colp
|
aa0fcfff6b |
chan_sip: Fix an issue where an incompatible audio format may be added to SDP.
If preferred codecs included any non-audio format the code would mistakenly add the audio format, even if it was not a joint capability with the remote side. (closes issue ASTERISK-21131) Reported by: nbougues Patches: patch_unsupported_codec_1.8.patch uploaded by nbougues (license 6470) ........ Merged revisions 401497 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Matthew Jordan
|
0336f4b606 |
res_rtp_asterisk: Fix crash when RTCP is not available during SSRC change
In r400089, a patch was put in to correct erroneous RTCP statistic resets. Unfortunately, ast_rtp_read can be called on an RTP instance that does not have RTCP information. This patch prevents that crash by only resetting the statistics if we do actually have an RTCP instance. (issue AST-1174) (closes issue ASTERISK-22667) Reported by: John Bigelow ........ Merged revisions 401445 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
2a40f6219a |
app_queue: Fix CLI "queue remove member" queue_log entry.
The queue_log entry resulting from CLI "queue remove member" when
log_membername_as_agent is enabled is wrong. It always uses the interface
name instead of the member name in the queue_log entry.
* Get the queue member before removing it from the queue so the member
name is available for the queue_log entry.
(closes issue ASTERISK-21826)
Reported by: Oscar Esteve
Patches:
fix_membername.diff (license #6505) patch uploaded by Oscar Esteve
(modified to fix potential ref leak)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Richard Mudgett
|
c1066265ad |
chan_dahdi: Fix unable to get index warning when transferring an analog call.
Transferring an analog call using flashhooks generated an unable to get index WARNING message when the transfer is completed. * Removed unnecessary analog subchannel shell games when transferring a call using flashhooks. Thanks to Tzafrir Cohen for mentioning this in a comment on issue ASTERISK-22720. ........ Merged revisions 401378 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401379 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kevin Harwell
|
a1ddd8987d |
Segfault in LIBEDIT_INTERNAL after tgetstr(), when libncurses5-dev
isn't installed
Include the appropriate declarations when not using termcap, but term+curses
and [n]curses do not exist.
(closes issue ASTERISK-22351)
Reported by: A. Iglesias
Patches:
issueA22351_libedit_internal_without_ncurses_dev.patch uploaded by wdoekes (license 5674)
........
Merged revisions 401325 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
Michael L. Young
|
44826ac192 |
Remove Port Restriction When Checking For NAT
When trying to determine if a peer is behind NAT, we should not be using the
ports when comparing addresses.
This patch removes the port from being checked and just useds the addresses
now.
(closes issue ASTERISK-22729)
Reported by: Michael L. Young
Tested by: Michael L. Young
Patches:
asterisk-remove-using-port-for-nat-check.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2927/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
Walter Doekes
|
8a6db34dc8 |
Properly copy/remove the device state cache flag over a masquerade.
In r378303 the AST_FLAG_DISABLE_DEVSTATE_CACHE flag was added that tells the devstate system to not cache states for non-real devices. However, when optimizing away channels (ast_do_masquerade), that flag wasn't copied. In my case, using Local devices as queue members created a situation where the endpoint was considered in use, but the state change of the device being available again was ignored (not cached). The endpoint channel was optimized into the (previously) Local channel, but kept the do-not-cache flag. The end result being that the queue member apparently stayed in use forever. (closes issue ASTERISK-22718) Reported by: Walter Doekes Review: https://reviewboard.asterisk.org/r/2925/ ........ Merged revisions 401178 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401179 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Michael L. Young
|
5ee5dccf51 |
Fix Setting A chan_sip Dialog's SIP_NAT_FORCE_RPORT Flag
A condition was added in a commit to fix ASTERISK-21374, that, if the
SIP_PAGE3_NAT_AUTO_RPORT flag was set, to then copy a peer's SIP_NAT_FORCE_RPORT
flag to the dialog. This condition should not have been there since it assumed
that if Asterisk is in an environment where NAT is involved, that the auto_* nat
settings or force_rport setting would be on in the global settings. If the nat
setting in the global setting is set to 'nat=no' and then turned on for peers
(which is not quite the recommended way, although it is allowed) this flag is
never copied to the dialog resulting in problems like, REGISTER replies going
to the wrong port.
This patch removes this conditional check and will now always use the peer's
flag which by this point in the code the checks on whether the peer is behind
NAT or not (if using auto_force_rport) have already been run.
(closes issue ASTERISK-22236)
Reported by: Filip Frank
Tested by: Michael L. Young
Patches:
asterisk-2236-always-set-rport.diff uploaded
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2919/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Kinsey Moore
|
75ec0df0a0 |
Reduce log level of a non-pubsub error message
Drop an error log message to debug level 1 since distributed device
state functions correctly when receiving this message and it spams the
logs.
(closes issue ASTERISK-22410)
Reported by: abelbeck
Patches:
asterisk-1.8-res_jabber-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
asterisk-11-res_xmpp-log-nonpubsub-error-to-debug.patch uploaded by abelbeck (License 5903)
........
Merged revisions 401119 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Walter Doekes
|
20b41f41c2 |
Don't check all realtime queues when doing "queue show some_queue".
When using realtime queues, queues have to be fetched from the database every now and then to see if any info has been changed or to see if the queue has been removed. When fetching info for an individual queue, the pruning of other queues is unnecessarily costly. Review: https://reviewboard.asterisk.org/r/2907/ ........ Merged revisions 401049 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401076 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
82a3fa8032 |
chan_iax2: Fix channel left locked in off nominal code path.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@401016 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Mark Michelson
|
ea53dd63be |
Prevent chan_sip from sending duplicate BYEs.
When a 200 OK for an initial INVITE is received, we were doing the right thing by ACKing and sending an immediate BYE. However, we also were doing the wrong thing and queuing an answer frame, thus causing the call to be answered. This would cause the call to be hung up by the channel thread, thus resulting in a second BYE being sent out. In this fix, I also have set the hangupcause to be correct since the initial BYE being sent by Asterisk had an unknown hangup cause. I have changed to using "Bearer capabilty not available" since the call was hung up due to an SDP offer/answer error. (closes issue ASTERISK-22621) reported by Kinsey Moore ........ Merged revisions 400970 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400971 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
f46a73f1c4 |
chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" output.
* Remember the swgain setting from CLI "dahdi set swgain" command so the
CLI "dahdi show channel" output will reflect the current setting.
* Updated CLI "dahdi set hwgain" and "dahdi set swgain" documentation.
(issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded by rmudgett
........
Merged revisions 400907 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400909 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Mark Michelson
|
23b6a34d45 |
chan_sip: Do not increment the SDP version between 183 and 200 responses.
Bumping the SDP version number can cause interoperability problems since receivers of the responses will expect that a 200 SDP will be identical to a previous 183 SDP. (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches: dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418) ........ Merged revisions 400906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400908 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
688f76dffb |
Add warning when compiling with iODBC support
When running configure, libiodbc2 development headers will fulfill the
requirement for ODBC development headers, but will not function
properly. This adds a warning when libiodbc2 development headers are
detected instead of unixodbc development headers.
(closes issue ASTERISK-22459)
Reported by: Patrick Maille
Tested by: Walter Doekes
Patches:
issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes (License 5674)
........
Merged revisions 400767 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Richard Mudgett
|
ae78f04e4f |
app_confbridge: Can now set the language used for announcements to the conference.
ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in
confbridge.conf or by the dialplan function
CONFBRIDGE(bridge,language)=en.
(closes issue ASTERISK-19983)
Reported by: Jonathan White
Patches:
M19983_rev2.diff (license #5138) patch uploaded by junky (modified)
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Richard Mudgett
|
3bc28a1af4 |
app_confbridge: Fix duplicate default_user profile.
* Fixed looking in the wrong profiles container to see if the default_user profile is already created in verify_default_profiles(). The bridge profile container is never going to hold user profiles. :) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400723 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
09df02a03e |
Fix func_config list entry allocation
The AST_CONFIG dialplan function defined in func_config.c allocates its config file list entries using ast_malloc. List entry allocations destined for use with Asterisk's linked list API must be ast_calloc()d or otherwise initialized so that list pointers are set to NULL. These uses of ast_malloc have been replaced by ast_calloc to prevent dereferencing of uninitialized pointer values when traversing the list. (closes issue ASTERISK-22483) Reported by: Brian Scott ........ Merged revisions 400694 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
69ab5a7cd1 |
Fix STUN crash when using IPv6 any address
Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4
candidates are also added.
(closes issue ASTERISK-21917)
Reported by: Torrey Searle
Patches:
0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400681 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
13 years ago |
|
Michael L. Young
|
ac850376a5 |
app_queue: Fix Queuelog EXITWITHKEY only logging two of four fields
Commit r62462 added two extra fields for logging "the original position the
caller entered the queue at, and the amount of time the caller was waiting in
the queue." But when r75969 was merged from 1.4 into trunk (r75977), these two
fields disappeared. Those two extra fields were not logged in 1.4 and when the
patch was merged, those fields went away.
Therefore, this is a regression and was caught by the reporter because he was
reading the awesome "Asterisk: The Definitive Guide" book.
(closes issue ASTERISK-22197)
Reported by: Dalius M.
Tested by: Dalius M.
Patches:
asterisk-22197-q-log-exitwithkey.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2901/
........
Merged revisions 400622 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Jonathan Rose
|
b8980c3be8 |
chan_sip: Don't ignore expires value in contact header if it lacks semicolon
(closes issue ASTERISK-22574)
Reported by: Filip Jenicek
Patches:
chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
........
Merged revisions 400469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Kinsey Moore
|
a055bdd4b4 |
Fix security events for AMI invalid password
In r337595, additional security events were added for chan_sip authentication failures. The new IEs added to the existing invalid password event were defined as required IEs, but existing users of the event did not set the new IEs and could not since they didn't apply to existing uses. They are now marked as optional IEs. (closes issue ASTERISK-22578) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400421 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
7652756518 |
res_rtp_multicast: Ensure SSRC is set properly
This fixes a bug where the SSRC field on multicast RTP can be stuck at
0 which can cause problems for endpoints trying to make sense of
incoming streams.
(closes issue ASTERISK-22567)
Reported by: Simone Camporeale
Patches:
22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536)
........
Merged revisions 400393 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
|
Michael L. Young
|
b4e0f80ec0 |
Cast Integer Argument To Unsigned Char
The member reg in the peercnt structure is an unsigned char and peercnt_modify() is expecting an unsigned char argument which gets assigned to peercnt->reg. This patch fixes that by casting the integer argument being passed to peercnt_modify to unsigned char. ........ Merged revisions 400314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400315 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
Tzafrir Cohen
|
093625b9fd |
man pages for astdb2bdb and astdb2sqlite3
Review: https://reviewboard.asterisk.org/r/2898/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400279 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Kinsey Moore
|
fb1c96baf4 |
chan_sip: Allow Asterisk to retry after 403 on register
This adds a global option in chan_sip to allow it to continue attempting registration if a 403 is received, clearing the cached nonce and treating it as a non-fatal response. Normally, this would cause registration attempts to that endpoint to stop. (closes issue ASTERISK-17138) Review: https://reviewboard.asterisk.org/r/2874/ Reported by: Rudi ........ Merged revisions 400137 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400140 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
ecdd1e76eb |
res_rtp_asterisk: Correct erroneous lost packet information in RTCP reports
RTCP's calculation of the number of lost packets in an RTP stream is based on that stream's sequence number count, the number of received packets, and how many packets we expect to receive. When the SSRC for an RTP stream changes, there can - and almost always will be - a large jump in the next packet's timestamp and sequence number. If we don't reset the number of received packets, sequence number count, and other metrics used by RTCP, the next RR/SR report will use the previous SSRC's values to calculate the lost packet count for the new SSRC - resulting in a very large number of lost packets. This patch modifies res_rtp_asterisk such that, if it detects a SSRC change, it will reset the various values used by the RTCP calculations. From the perspective of RTCP, this appears as a new media stream - which is what it is. Review: https://reviewboard.asterisk.org/r/2886/ (closes issue AST-1174) Reported by: Thomas Arimont ........ Merged revisions 400089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400093 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Matthew Jordan
|
4886cd2d37 |
Add check for openSUSE when detecting bfd library
In ASTERISK-17842, some additional library checks were added to the configure script so that the bfd library could be found on CentOS and Fedora systems. As it turns out, openSUSE requires an additional library. This patch adds another check to the configure script for openSUSE that will add that library. Review: https://reviewboard.asterisk.org/r/2885/ (closes issue AST-1169) Reported by: Guenther Kelleter ........ Merged revisions 400073 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400075 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
ee37433d8e |
chan_sip: Increase some scratch buffer sizes dealing with caller id.
* Eliminated an unnecessary initialization in check_user_full(). (closes issue ASTERISK-22477) Reported by: Michael Shepelev ........ Merged revisions 400013 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@400014 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Jonathan Rose
|
b9133abc09 |
chan_sip: Reject calls on 200 OKs if no SDP has been received
When Asterisk receives a 200 OK in response to an invite, that peer should have sent an SDP at some point by then. If the channel has never received an SDP, media won't have been set and the remote address won't be known. Endpoints in general should not be doing this. This patch makes it so that Asterisk will simply hang up a call if it sends a 200 OK at this point. So far this odd behavior for endpoints has only been observed in tests which involved manually created SIP transactions in SIPp. (closes issue ASTERISK-22424) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2827/ ........ Merged revisions 399939 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
Richard Mudgett
|
7ca3e44c3f |
chan_dahdi: CLI "core stop gracefully" has needless delay for PRI and SS7.
The PRI and SS7 link control threads are not stopped correctly when the chan_dahdi.so module is unloaded. The link control threads pri_dchannel() and ss7_linkset() are not awakened from a poll() to cancel the thread. * Added a SIGURG signal after requesting the thread cancel to break the link control thread poll() immediately. For SS7 it was slightly worse, the link poll() timeout would always be whatever was the last libss7 scheduled event time used. If no libss7 scheduled event was pending, the thread could run more often than necessary. * Set nextms to 60 seconds for the ss7_linkset() poll() if there is no other libss7 scheduled event. ........ Merged revisions 399818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Michael L. Young
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591c3b403e |
chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok
1st Issue
When a realtime peer sends an un-REGISTER request, Asterisk
un-registers the peer but the database table record still has regseconds and
fullcontact for the peer. This results in calls attempting to be routed to the
peer which is no longer registered. The expected behavior is to get
busy/congested when attempting to call an un-registered peer through the
dialplan.
What was discovered is that we are clearing out the peer's registration in the
database in parse_register_contact() when calling expire_register() but then
upon returning from parse_register_contact(), update_peer() is run which stores
back in the database table regseconds and fullcontact.
2nd Issue
The reporter pointed out that the 200 ok being returned by Asterisk
after un-registering a peer contains a Contact header with ;expires= and the
Expires header is not set to 0. This is actually a regression.
Tests were created for this second issue (ASTERISK-22548). The tests have been
reviewed and a Ship It! was received on those tests.
This patch does the following:
* Do not ignore the Expires header value even when it is set to 0. The patch
sets the pvt->expiry earlier on in the function so that it is set properly and
used.
* If pvt->expiry is 0, do not call update_peer since that means the peer has
already been un-registered and there is no need to update the database record
again since nothing has changed.
(closes issue ASTERISK-22428)
Reported by: Ben Smithurst
Tested by: Ben Smithurst, Michael L. Young
Patches:
asterisk-22428-rt-peer-update-and-expires-header.diff
by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2869/
........
Merged revisions 399794 from http://svn.asterisk.org/svn/asterisk/branches/1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399795 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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13 years ago |
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Richard Mudgett
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46c2c5ea14 |
chan_iax2: Prevent some needless breaking of the native IAX2 bridge.
* Clean up some twisted code in the iax2_bridge() loop. * Add AST_CONTROL_VIDUPDATE and AST_CONTROL_SRCCHANGE to a list of frames to prevent the native bridge loop from breaking. * Passing the AST_CONTROL_T38_PARAMETERS frame should also allow FAX over a native IAX2 bridge. (issue ABE-2912) Review: https://reviewboard.asterisk.org/r/2870/ ........ Merged revisions 399697 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399708 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Kinsey Moore
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2fb3e52fda |
Ensure global types in the config framework are initialized
If a config object was allocated but one of its global objects was never encountered, then the global object's defaults were never applied. Ensure that global objects are initialized properly upon allocation instead of on configuration. Review: https://reviewboard.asterisk.org/r/2866/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399564 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Kevin Harwell
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44f616bd2e |
Fix memory leak in logger.
Fixed a memory leak discovered in the logger where a temporary string buffer was not being freed. (closes issue ASTERISK-22540) Reported by: John Hardin git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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Jonathan Rose
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155e9f821c |
chan_sip: Make direct media reinvites for T38 put Asterisk in the media path
Prior to this patch, Asterisk would incorrectly use the previous endpoint addresses in SDP in spite of providing its own port. T38 is never meant to be done through directmedia and Asterisk should always be in the media path for these streams. (closes issue ASTERISK-17273) Reported by: Kevin Stewart (closes issue ASTERISK-18706) Reported by: Jeremy Kister Review: https://reviewboard.asterisk.org/r/2853/ ........ Merged revisions 399456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@399457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |