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23318 Commits (dc06b355474390c7d94106330aad89f275984c24)
Author | SHA1 | Message | Date |
---|---|---|---|
|
af1666f1cf |
Reset RTP timestamp; sequence number on SSRC change
In r370252 for ASTERISK-18404, Asterisk's handling of RTP was modified to better account for out of order RTP packets. This was accomplished by using the RTP timestamp and sequence number to check for out of order packets. However, when a SSRC change occurs, the timestamp and sequence number will no longer have any relation to the previously received packets. The variables tracking the timestamp and sequence number therefore have to be reset. (closes issue ASTERISK-20906) Reported by: Eelco Brolman patches: dtmf_on_hold.patch uploaded by Eelco Brolman (license #6442) ........ Merged revisions 378967 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378984 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
12c51024c3 |
Fix XML encoding of 'identity display' in NOTIFY messages.
XML encoding in chan_sip is accomplished by naively building the XML directly from strings. While this usually works, it fails to take into account escaping the reserved characters in XML. This patch adds an 'ast_xml_escape' function, which works similarly to 'ast_uri_encode'. This is used to properly escape the local_display attribute in XML formatted NOTIFY messages. Several things to note: * The Right Thing(TM) to do would probably be to replace the ast_build_string stuff with building an ast_xml_doc. That's a much bigger change, and out of scope for the original ticket, so I refrained myself. * It is with great sadness that I wrote my own ast_xml_escape function. There's one in libxml2, but it's knee-deep in libxml2-ness, and not easily used to one-off escape a string. * I only escaped the string we know is causing problems (local_display). At least some of the other strings are URI-encoded, which should be XML safe. Rather than figuring out what's safe and escaping what's not, it would be much cleaner to simply build an ast_xml_doc for the messages and let the XML library do the XML escaping. Like I said, that's out of scope. (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by: Guenther Kelleter Review: http://reviewboard.digium.internal/r/365/ ........ Merged revision 378919 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 378933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378934 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
cef6475eb0 |
Retain XMPP filters across reconnections so external modules continue to function as expected.
Previously if an XMPP client reconnected any filters added by an external module were lost. This issue exhibited itself with chan_motif not receiving and reacting to Jingle signaling. (closes issue ASTERISK-20916) Reported by: kuj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
48b3e46f00 |
Fix end condition in ast_rtp_lookup_mime_multiple2.
The erroneous end condition would never include the AST_RTP_CISCO_DTMF flag in the debug output. (closes issue ASTERISK-20772) Reported by: Xavier Hienne ........ Merged revisions 378776 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378780 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
1f6610acec |
Move declaration of ast_regex_string_to_regex_pattern futher down strings.h.
The prior location is before the declaration of struct ast_str, which causes compiler warnings. (closes issue ASTERISK-20852) Reported by: Pavel Troller Patches: strings.diff uploaded by Pavel Troller (license 6302) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378747 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
8d7224d5a8 |
Replace errant tabs with spaces in causes.h.
(closes issue ASTERISK-20826) Reported by: snuffy Patches: notabs.dif uploaded by snuffy (license 5024) ........ Merged revisions 378733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378734 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
0317ece036 |
app_queue: Fix incorrect assertion.
(issue ASTERISK-16115) ........ Merged revisions 378689 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378690 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
69fc13fb67 |
app_queue: Fix multiple calls to a queue member that is in only one queue.
When ringinuse=no queue members can receive more than one call if these calls happen at nearly the same time. * Fix so a queue member does not receive more than one call from a queue. NOTE: This fix does not prevent multiple calls to a member if the member is in more than one queue. * Did some refactoring to eliminate some code redundancy. (issue ASTERISK-16115) Reported by: nik600 Patches: jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett Modified * Revert the -r341580 and -r341599 changes adding the queues.conf check_state_unknown option as it was added in an attempt to fix this problem. The fix did not need to be optional. The fix should not have tried to explicitly set the device state. Setting the device state by something other than the device introduces a race condition. I also could not see how the change would be effective other than delaying the app_queue code long enough for the device state to propagate to app_queue. ........ Merged revisions 378663 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378683 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378687 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
d35f6c5210 |
Rewrite skinny dialing to remove threaded simpleswitch
This rewrite changes skinny dialing from the threaded simpleswitch to a scheduled timeout approach. There were some underlying issues with the threaded simple switch with occasional corruption and possible segfaults. Review: https://reviewboard.asterisk.org/r/2240/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
d566b201ae |
res_srtp: Prevent a crash from occurring due to srtp_create failures in srtp_create
Under some circumstances, libsrtp's srtp_create function deallocates memory that it wasn't initially responsible for allocating. Because we weren't initially aware of this behavior, this memory was still used in spite of being unallocated during the course of the srtp_unprotect function. A while back I made a patch which would set this value to NULL, but that exposed a possible condition where we would then try to check a member of the struct which would cause a segfault. In order to address these problems, ast_srtp_unprotect will now set an error value when it ends without a valid SRTP session which will result in the caller of srtp_unprotect observing this error and hanging up the relevant channel instead of trying to keep using the invalid session address. (closes issue ASTERISK-20499) Reported by: Tootai Review: https://reviewboard.asterisk.org/r/2228/diff/#index_header ........ Merged revisions 378591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378592 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
8653377baf |
Fix pjproject compilation in certain circumstances
On a fresh checkout of Asterisk 11, running make before ./configure could cause the pjproject subdirectory to get in an odd state that would prevent compilation. This patch by Tilghman prevents that from occurring. (closes issue ASTERISK-20681) Reported by: Dinesh Ramjuttun Tested by: danilo borges, Steve Lang patches: 20121208__ccar_solved.diff.txt uploaded by Tilghman Lesher (license 5003) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378582 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
10a948a8ca |
Fix SIP Notify Messages To Have The Proper IP Address In The FROM Field
On a multihomed server when sending a NOTIFY message, we were not figuring out which network should be used to contact the peer. This patch fixes the problem by calling ast_sip_ouraddrfor() and then build_via() so that our NOTIFY message contains the correct IP address. Also, a debug message is being added to help follow the call-id changes that occur. This was helpful for confirming that the IP address was set properly since the call-id contains the IP address. It also will be helpful for troubleshooting purposes when following a call in the debug logs. (closes issue ASTERISK-20805) Reported by: Bryan Hunt Tested by: Bryan Hunt, Michael L. Young Patches: asterisk-20805-notify-ip-v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2255/ ........ Merged revisions 378554 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
c4587d7170 |
Don't pass STUN packets through the SRTP unprotect function.
(closes issue AST-1036) Reported by: jbigelow ........ Merged revisions 378553 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378555 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
aeeeb28d27 |
Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present
When the "h" extension is present within the context of the queue, all calls are being reported COMPLETECALLER even when the agent is hanging up the call. This patch checks to see if the agent hung-up or not instead of only relying on checking if the queue (caller) channel hung-up or not. It would appear that having the h extension in the mix, the pbx goes to the h extension, "hanging-up" the queue channel and triggering the reporting of COMPLETECALLER. (closes issue ASTERISK-20743) Reported by: call Tested by: call, Michael L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2256/ ........ Merged revisions 378514 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378515 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
f6ae419786 |
chan_agent: Fix wrapup time wait response.
* Made agent_cont_sleep() and agent_ack_sleep() stop waiting if the wrapup time expires. agent_cont_sleep() had tried but returned the wrong value to stop waiting. * Made agent_ack_sleep() take a struct agent_pvt pointer instead of a void pointer for better type safety. ........ Merged revisions 378486 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378487 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
914397aa23 |
Add missing test event
This test event was missing from channel.c causing the dial_LS_options test to fail intermittently because of a race condition where most code paths emitted the test event but this one did not. The dial_LS_options test should stop bouncing now. ........ Merged revisions 378455 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378459 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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ce8d64cd99 |
chan_agent: Misc code cleanup.
* Fix off-nominal path resource cleanup in agent_request(). * Create agent_pvt_destroy() to eliminate inlined versions in many places. * Pull invariant code out of loop in add_agent(). * Remove redundant module user references in login_exec(). * Remove unused struct agent_pvt logincallerid[] member. * Remove some redundant code in agent_request(). ........ Merged revisions 378456 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378457 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
bba3c7a1f2 |
chan_agent: Fix agent_indicate() locking.
Avoid deadlock potential with local channels and simplify the locking. ........ Merged revisions 378427 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378428 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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17467a6926 |
Prevent exhaustion of system resources through exploitation of event cache
This patch changes res_xmpp to no longer cache events under certain circumstances. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378411 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
3778ed7fc1 |
Prevent crashes in res_xmpp when receiving large messages
Similar to r378287, res_xmpp was marshaling data read from an external source onto the stack. For a sufficiently large message, this could cause a stack overflow. This patch modifies res_xmpp in a similar fashion to res_jabber by removing the stack allocation, as it was unnecessary. (issue ASTERISK-20658) Reported by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378409 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
7eb9a846d8 |
Prevent crashes from occurring when reading from data sources with large values
When reading configuration data from an Asterisk .conf file or when pulling data from an Asterisk RealTime backend, Asterisk was copying the data on the stack for manipulation. Unfortunately, it is possible to read configuration data or realtime data from some data source that provides a large blob of characters. This could potentially cause a crash via a stack overflow. This patch prevents large sets of data from being read from an ARA backend or from an Asterisk conf file. (issue ASTERISK-20658) Reported by: wdoekes Tested by: wdoekes, mmichelson patches: * issueA20658_dont_process_overlong_config_lines.patch uploaded by wdoekes (license 5674) * issueA20658_func_realtime_limit.patch uploaded by wdoekes (license 5674) ........ Merged revisions 378375 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378376 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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b1366d7acf |
Fix AMI redirect action with two channels failing to redirect both channels.
The AMI redirect action can fail to redirect two channels that are bridged together. There is a race between the AMI thread redirecting the two channels and the bridge thread noticing that a channel is hungup from the redirects. * Made the bridge wait for both channels to be redirected before exiting. * Made the AMI redirect check that all required headers are present before proceeding with the redirection. * Made the AMI redirect require that any supplied ExtraChannel exist before proceeding. Previously the code fell back to a single channel redirect operation. (closes issue ASTERISK-18975) Reported by: Ben Klang (closes issue ASTERISK-19948) Reported by: Brent Dalgleish Patches: jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/ ........ Merged revisions 378356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378358 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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4ae3e66995 |
Restore branch-1.8-merged on 11
This was accidentally deleted during a merge. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378337 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
|
eda6664de0 |
Prevent exhaustion of system resources through exploitation of event cache
Asterisk maintains an internal cache for devices in the event subsystem. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include entities that do not have a physical representation. One way that this occurred was when anonymous calls are allowed in Asterisk. A device was automatically created and stored in the cache for each anonymous call that occurred; this was possible in the SIP and IAX2 channel drivers and through channel drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices are never removed from the system, allowing anonymous calls to potentially exhaust a system's resources. This patch changes the event cache subsystem and device state management to no longer cache devices that are not associated with a physical entity. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore patches: event-cachability-3.diff uploaded by jcolp (license 5000) ........ Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378321 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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5ebec60090 |
Resolve crashes due to large stack allocations when using TCP
Asterisk had several places where messages received over various network transports may be copied in a single stack allocation. In the case of TCP, since multiple packets in a stream may be concatenated together, this can lead to large allocations that overflow the stack. This patch modifies those portions of Asterisk using TCP to either favor heap allocations or use an upper bound to ensure that the stack will not overflow: * For SIP, the allocation now has an upper limit * For HTTP, the allocation is now a heap allocation instead of a stack allocation * For XMPP (in res_jabber), the allocation has been eliminated since it was unnecesary. Note that the HTTP portion of this issue was independently found by Brandon Edwards of Exodus Intelligence. (issue ASTERISK-20658) Reported by: wdoekes, Brandon Edwards Tested by: mmichelson, wdoekes patches: ASTERISK-20658_res_jabber.c.patch uploaded by mmichelson (license 5049) issueA20658_http_postvars_use_malloc2.patch uploaded by wdoekes (license 5674) issueA20658_limit_sip_packet_size3.patch uploaded by wdoekes (license 5674) ........ Merged revisions 378269 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378286 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378287 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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28b36d4864 |
Ensure chan_sip rejects encrypted streams without crypto info
This ensures that Asterisk rejects encrypted media streams (RTP/SAVP audio and video) that are missing cryptographic keys and ensures that the incoming SDP is consistent with RFC4568 as far as having a crypto attribute present for any SAVP streams. Review: https://reviewboard.asterisk.org/r/2204/ ........ Merged revisions 378217 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378218 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378219 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c19b6731ba |
Give the causes[] a struct name.
........ Merged revisions 378164 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378165 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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82ac496986 |
Add branch-1.8-merged property to allow direct merging from v1.8
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378163 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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32e7992bd5 |
Add test events for time limit-related hangups
This patch adds hangup-related test events in order to support testing of time-limited bridges. This aids in testing the S() and L() bridge options. (issue SWP-4713) ........ Merged revisions 378119 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378120 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378121 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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d571d53fa4 |
Fix potential double free when unloading a module.
........ Merged revisions 378092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378093 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378094 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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ac78bb107a |
Make chan_local module references tied to local_pvt lifetime.
The chan_local module references were manually tied to the existence of the ;1 and ;2 channel links. * Made chan_local module references tied to the existence of the local_pvt structure as well as automatically take care of the module references. * Tweaked the wording of the local_fixup() failure warning message to make sense. Review: https://reviewboard.asterisk.org/r/2181/ ........ Merged revisions 378088 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378089 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378090 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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c2f104641c |
Make libasteriskssl.so symlink use a relative path.
This was causing issues when using DESTDIR, since the path to which the link pointed is not likely to exist (and not useful to exist) on the target system. (issue ASTNOW-284) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378073 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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e623bdcea8 |
app_queue: Revert bad ringinuse=no patch.
With the option ringinuse=no set, the patch committed for ASTERISK-16115 causes non-SIP queue members to never be called because the device state is checked after a channel is created to determine if the member is busy. These queue members always get the "Member %s is busy, cannot dial" message. Most channel drivers other than chan_sip use the default device state handling. The default device-state state is considered in use or unknown if the channel exists or not respectively. (closes issue ASTERISK-20801) Reported by: rmudgett Patches: jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 378036 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378037 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378038 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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62007e30f8 |
Fix skinny to recognise vmexten in general section of conf
Fixup the vmexten so if globally set in general section will be honored by chan_skinny. Also get rid of the 'global_' part of variable name to match regexten. (closes issue ASTERISK-20790) Reported by: snuffy Tested by: snuffy, myself Patches: skinny-vm.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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797e403982 |
confbridge: Fix MOH on simultaneous user entry to a new conference.
When two users entered a new conference simultaneously, one of the callers hears MOH. This happened if two unmarked users entered simultaneously and also if a waitmarked and a marked user entered simultaneously. * Created a confbridge internal MOH API to eliminate the inlined MOH handling code. Note that the conference mixing bridge needs to be locked when actually starting/stopping MOH because there is a small window between the conference join unsuspend MOH and actually joining the mixing bridge. * Created the concept of suspended MOH so it can be interrupted while conference join announcements to the user and DTMF features can operate. * Suspend any MOH until the user is about to actually join the mixing bridge of the conference. This way any pre-join file playback does not need to worry about MOH. * Made post-join actions only play deferred entry announcement files. Changing the user/conference state during that time is not protected or controlled by the state machine. (closes issue ASTERISK-20606) Reported by: Eugenia Belova Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2232/ ........ Merged revisions 377992 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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d3ab29934c |
Minor fixes for chan_skinny
Whitespace, change SUBSTATE_ONHOOK to correct SKINNY_ONHOOK and correct len of 2 strcmp in skinny_setdebug(). (see opticron's review on https://reviewboard.asterisk.org/r/2240/) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377991 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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38d06ca3d2 |
Fix skinny debug tab completion
Review the syntax of the 'skinny debug' command to show more than just 'show' for options to 'skinny debug' command. (closes issue ASTERISK-20789) Reported by: snuffy Tested by: snuffy, myself Patches: skinny-debug.diff uploaded by snuffy (license 5024) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377985 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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3292e66ca1 |
Ensure Min-SE is included in outbound INVITEs
Asterisk now includes Min-SE in outbound INVITEs when the value is not 90 (the default) and session timers are not disabled. This has the effect of Asterisk following RFC4028 more closely with regard to 422 responses and preventing situations in which Asterisk would be forced to temporarily accept a call to tear it down based on a Session-Expires below the locally configured Min-SE. (issue SWP-5051) Review: https://reviewboard.asterisk.org/r/2222/ Reported-by: Kinsey Moore Patch-by: Kinsey Moore ........ Merged revisions 377946 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377947 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377948 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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990a2e764d |
Incremented EXTRA_SOUNDS_VERSION in sounds/Makefile to 1.4.12 for new Extra Sounds releases
See CHANGES-* files in English extra 1.4.12 tarballs for new sound prompts added. (closes ASTERISK-20328) Reported by: Matt Jordan (closes AST-755) Reported by: John Bigelow ........ Merged revisions 377922 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377923 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377924 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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b6d36124cd |
Fix a potential deadlock in chan_sip during transfers.
The issue comes from the fact that transfers may perform a redirecting update on a channel. The issue is that lock inversion between the channel and its tech_pvt occurs since the channel lock is released during the transfer process. The fix is to move when the redirecting update occurs to a place where neither the tech_pvt or the channel is locked so that the two can be locked in the proper order. (closes issue ASTERISK-20708) reported by Mark Michelson patches: ASTERISK-20708-3.patch uploaded by Mark Michelson (License #5049) Tested by: Tim Ringenbach at Asteria Solutions Group git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377910 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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573e76447c |
Cleanup CLI commands on exit for several files.
(issue ASTERISK-20649) Reported by: Corey Farrell Patches: unregister-cli-multiple-all.patch (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 377881 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377882 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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0d7879be8c |
Cleanup udptl on exit.
* Cleanup CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: udptl-shutdown-1_8-10.patch (license #5909) patch uploaded by Corey Farrell udptl-shutdown-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377848 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377849 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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bdd5921950 |
Fix crash that can occur if CLI registration fails for an aliased command.
A recent memory leak fix in main/cli.c causes an ast_cli_entry's command field to be freed and NULLed if ast_cli_register() fails. res_clialiases was ignoring the return value of ast_cli_register() and was then passing the NULL command off to a a hash function. This resulted in a crash. The fix is not to ignore the erroneous return value. If ast_cli_register() fails, then we do not continue trying to process the current alias. ........ Merged revisions 377840 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377842 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377843 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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920da001d8 |
Cleanup taskprocessor on exit.
* Cleanup CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: taskprocessor-cleanup-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell taskprocessor-cleanup-10-only.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377837 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377838 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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737d2e55c6 |
Cleanup pbx on exit.
* Cleanup CLI commands on exit. * Unreference hints and statecbs containers on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: pbx-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell pbx-cleanup-10.patch (license #5909) patch uploaded by Corey Farrell pbx-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377806 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377807 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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00ea66650b |
Cleanup logger on exit.
* Cleanup CLI commands, destroy verbosers and logchannels lists on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: logger-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377771 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377772 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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d73bad943d |
Cleanup indications on exit.
* Made ast_unregister_indication_country() unlink the found tone zone before selecting a new default_tone_zone to make it impossible to select the tone zone being unregistered again. * Ringcadence is no longer parsed twice in store_config_tone_zone(). * Cleanup CLI commands and destroy default_tone_zone on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: indications-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377740 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377741 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377742 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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a65e904a51 |
Cleanup event on exit.
* Cleanup CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: event_shutdown-10-only.patch (license #5909) patch uploaded by Corey Farrell event_shutdown-1_8-11-trunk.patch (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 377708 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377709 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377710 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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6368a140a0 |
Cleanup dnsmgr on exit.
* Cleanup dnsmgr thread and CLI commands on exit. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: dnsmgr-cleanup-1_8.patch (license #5909) patch uploaded by Corey Farrell dnsmgr-cleanup-10-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377705 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377706 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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b2c2a84e17 |
Ensure ReceiveFax provides a CED tone via T.38
When using res_fax_digium, the T.38 CED tone was not being provided properly which would cause some incoming faxes to fail. This was not an issue with res_fax_spandsp since it does not strictly honor the send_ced flag and sends the CED tone whenever receiving a T.38 fax. (closes issue FAX-343) Reported-by: Benjamin Tietz Patch-by: Kinsey Moore ........ Merged revisions 377655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377656 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |