mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
14371 Commits (752c3ef59108932a8bec33af9504fb2b6fd93cfd)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Alec L Davis
|
752c3ef591 |
app_voicemail:close_mailbox imap_storage doesn't use last_msg_index
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Alec L Davis
|
c875064714 |
app_voicemail: close_mailbox needs to respect additional messages while mailbox is open.
close_mailbox leave gaps in message sequence if messages are deleted and new messages arrive during this time, this is because the shuffle down to slot 0, only shuffles the number of pre-existing messages when mailbox is opened, ignoring new arrivals. Fix: in close_mailbox re-evaluate number of messages before the shuffle, this then includes new arrivals. Happens on filebased or ODBC storage. (issues #19032,#18582,#18692,#18998) Reported by: alecdavis,tootai,afosorio Review: https://reviewboard.asterisk.org/r/1153/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@312070 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Matthew Nicholson
|
c0e4bdd609 |
Don't hold the pvt lock while streaming a file.
ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311203 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Matthew Nicholson
|
9dbd90f447 |
Don't dec the usecount of an eventqent then use it.
ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311201 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Matthew Nicholson
|
5510659c1f |
Remove the provisional keepalive scheduler entry's reference to the pvt when we remove the scheduler entry.
ABE-2756 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Alec L Davis
|
93d2033f4b |
Remove extra quote in indications.conf
Picking low hanging fruit. (closes issue #18971) Reported by: IgorG Patches: based on indications.conf.sample.diff uploaded by IgorG (license 20) Tested by: IgorG git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@311048 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Terry Wilson
|
bf061fa778 |
Don't delay DTMF in core bridge while listening for DTMF features
This patch is mostly the work of Olle Johansson. I did some cleanup and added the silence generating code if transmit_silence is set. When a channel listens for DTMF in the core bridge, the outbound DTMF is not sent until we have received DTMF_END. For a long DTMF, this is a disaster. We send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds. Some products see this delay and the time skew on RTP packets that results and start ignoring the audio that is sent afterward. With this change, the DTMF_BEGIN frame is inspected and checked. If it matches a feature code, we wait for DTMF_END and activate the feature as before. If transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a multi-digit feature. If it doesn't match a feature, the frame is forwarded along with the DTMF_END without delay. By doing it this way, DTMF is not delayed. (closes issue #15642) Reported by: jasonshugart Patches: issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396) Tested by: globalnetinc, jde (closes issue #16625) Reported by: sharvanek Review: https://reviewboard.asterisk.org/r/1092/ Review: https://reviewboard.asterisk.org/r/1125/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Alec L Davis
|
ba3c804b06 |
core show locks: display ThreadID in hexadecimal
Allow easier cross referencing of thread ID's with GDB backtraces (closes issue #18968) Reported by: alecdavis Patches: bug18968.diff.txt uploaded by alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310779 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Richard Mudgett
|
5e8dfa3e6b |
"Caller*ID failed checksum" on Wildcard TDM2400P and TDM410
The last character in the caller id message is getting a framing error. The checksum is the last character in the message. A framing error in the checksum could be because: 1) The sender did not send a full stop bit. 2) The sender cut off the FSK carrier too soon. 3) The sender opted to send zero of the specified zero to 10 trailing mark bits and round-off errors in the code resulted in the code not being where it thought it was in the demodulated bit stream. Bit 8 of 'b' is set when parity error. Bit 9 of 'b' is set when framing error. Made ignore the framing and parity error bits if the errored character is the checksum. We can tolerate a framing/parity error there. The checksum character validates the message. (closes issue #18474) Reported by: nivek Patches: callerid.c.1.patch uploaded by nivek (license 636) (with modifications) Tested by: nivek git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310633 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Tilghman Lesher
|
75dc421e15 |
Add AELSub, which provides a stable entry point into AEL subroutines.
This commit needs some explanation, given that we're adding a new application into an existing release branch. This is generally a violation of our release policy, except in very limited circumstances, and I believe this is one of those circumstances. The problem that this solves is one of the sanity of using multiple dialplan languages to define a dialplan. In the case of the reporter, he or she is using AEL is define subroutines, while using Realtime extensions to invoke those subroutines. While you can do this, it's based upon the reality of AEL using actual dialplan extensions; however, there is no guarantee that the details of _how_ AEL is compiled into extensions will remain stable. In fact, at the time of this commit, it has already changed twice, once in a fundamental way. Now normally, a new application would only be added to trunk. However, this application is explicitly to create a stable user-level API between versions, and adding it to trunk only will not solve the user's problem of switching between 1.6.2 and 1.8, nor will it help anybody switching from 1.8 to 1.10. Therefore, it needs to go into existing release branches. For the sake of consistency, and also because one of the changes was between 1.4 and 1.6.x, I am also electing to commit this to 1.4. (closes issue #18910) Reported by: alexandrekeller Patches: 20110304__issue18919__1.6.2.diff.txt uploaded by tilghman (license 14) 20110304__issue18919__1.4.diff.txt uploaded by tilghman (license 14) Tested by: alexandrekeller git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
|
eafdcbacd4 |
Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems.
(closes issue #18295) Reported by: pruiz git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@310140 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Terry Wilson
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744b5c4bbb |
Don't try to free statically allocated memory.
Note: compiling after ./configure --enable-dev-mode will keep these kinds of mistakes from being committed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@309947 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jonathan Rose
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aaf50c4688 |
Bug fix for MixMonitor involving filenames with '.' not in the extension
Closes issue #18391) Reported by: pabelanger Patches: bugfix.patch uploaded by jrose (license 1225) Tested by: jrose git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@309856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
David Ruggles
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fd2d1c2139 |
fix small memory leak
fix small memory leak caused by a string allocation that wasn't freed (closes issue #18907) Reported by: andy11 Patches: asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@309355 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Terry Wilson
|
e90101cecf |
Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection every time someone connects via AMI. It should only be sent to the user who just connected. (closes issue #18168) Reported by: FeyFre Patches: bug0018168.patch uploaded by FeyFre (license 1142) Tested by: FeyFre, twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308813 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Matthew Nicholson
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ebc809b6d4 |
silence gcc 4.2 compiler warning
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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2e714923c3 |
Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway.
AST-2011-002 FAX-281 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308413 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jason Parker
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788e66cf20 |
Fix regression that changed behavior of queues when ringing a queue member.
This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@308002 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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212f41ddf3 |
Reentrancy problem if outgoing call gets different B channel than requested.
The chan_dahdi pri_fixup_principle() routine needs to protect the Asterisk channel with the channel lock when it changes the technology private pointer to a new private structure. * Added lock protection while pri_fixup_principle() moves a call from one private structure to another. * Made some pri_fixup_principle() messages more meaningful. Partial backport from v1.8 -r300714. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@307623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jason Parker
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af36e55080 |
Remove color when executing commands via a remote console.
Essentially this makes '-x' imply '-n' on rasterisk. This was done in a different and incomplete way previously, which I'll be reverting shortly. (issue #18776) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@307534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
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9b490b1adc |
Fix comparison for REFER Replaces tags with pedantic=yes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306972 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Jeff Peeler
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d6eabdc0dd |
fix this line again
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306965 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Jeff Peeler
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fb475c8c7e |
clean this up, sorry my brain is not really working
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306964 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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8bd2bfb587 |
Backup file storing message duration is not used with IMAP_STORAGE, remove code.
The message duration is stored in the body of the email when using IMAP_STORAGE, so nothing needs to happen with the backup file. (closes issue #18718) Reported by: kerframil git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306960 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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e9dcd4a9e2 |
make this safer and fully correct, pointed out by Steve Davis
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
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dceced3626 |
Don't try to pickup a call in the middle of a masquerade
If A calls B which doesn't answer and C & D both try to do a call pickup, it is possible for ast_pickup_call to answer the call, then fail to masquerade one of the calls because the other one is already in the process of masquerading. This patch checks to see if the channel is in the process of masquerading before call before selecting it for a pickup. Review: https://reviewboard.asterisk.org/r/1094/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
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c80738307a |
Don't allow a REFER w/replaces to replace its own dialog
Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces header that matches the dialog of the REFER. This would be a situation like A calls B, A calls C, A transfers B to A, which is just silly. This patch makes the transfer fail instead of making Asterisk freak out and forget to hang other channels up. Review: https://reviewboard.asterisk.org/r/1093/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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a7d12781c0 |
Fix no MOH and frame queueing problem for parked calls.
This was a regression introduced when select was changed to poll and was just a conversion error: POLLPRI detects OOB data, not POLLERR. (closes issue #18637) Reported by: jvandal git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306120 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
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a5955e4b8f |
Set hangup cause in local_hangup
When a call involves a local channel (like SIP -> Local -> SIP), the hangup cause was not being set. This resulted in SIP channels sometimes getting a 503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+ this also can cause issues with CCSS that involve a local channel. This patch sets the hangupcause for one side of the local channel to the other in local_hangup for outbound calls. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@306119 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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9fe4bc60b5 |
Minor AST_FRAME_TEXT related issues.
* Include the null terminator in the buffer length. When the frame is queued it is copied. If the null terminator is not part of the frame buffer length, the receiver could see garbage appended onto it. * Add channel lock protection with ast_sendtext(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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5d3d66c13b |
Close file descriptor for timing source when a MOH class gets destroyed.
(closes issue #18457) Reported by: mcallist Patches: 18457-closetimer.diff uploaded by qwell (license 4) 18457-closetimer_trunk.diff uploaded by qwell (license 4) Tested by: qwell, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305471 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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46d7f5d734 |
Obtain the pri lock for PRI queue counters.
Need to obtain the pri lock when calling pri_dump_info_str() to avoid a reentrancy problem when calculating the Q.921 Q count statistic. JIRA AST-484 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305341 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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e03c6ed738 |
Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/))
chan_iax2 and other channel drivers already had code to prevent this. The attempt that app_dial was making to prevent it was not correct, so I fixed that. (closes issue #18371) Reported by: gbour Patches: 18371.patch uploaded by gbour (license 1162) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305252 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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23a1e7bb49 |
Set file descriptors to -1 on creation, so that we don't see weirdness later.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@305129 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
Tilghman Lesher
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3c5fd20e90 |
Fix compilation when ODBC_STORAGE is defined.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Sean Bright
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1fb86f03c8 |
Backport MeetMe related reference leaks fixes from 1.6.2/1.8/trunk.
I had forgotten that MeetMe in 1.4 also used astobj2, so backport the fixes where appropriate. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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0b416c5547 |
Fix default prefix=/usr regression on non-Linux systems.
This partially reverts a change made in branches/1.4/ r267759, which will cause issue #17013 to be reopened. This issue was pointed out by a user on #asterisk, who helpfully discovered that paths were being set incorrectly. To truly understand what was wrong, one should run: svn diff --force -c<this revision> configure git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304464 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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11720dff43 |
Rerun bootstrap.sh with no changes, so that it is more obvious what my next commit changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304460 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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0e62797165 |
Convert from network to host byte ordering before checking if an IP is a multicast address.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304247 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Mark Michelson
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13d35425dc |
Get rid of unused 'verbose' field in ast_udptl
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304242 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Matthew Nicholson
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a40495fe84 |
This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header.
ABE-2664 Review: https://reviewboard.asterisk.org/r/1059/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Sean Bright
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5a033015c5 |
Make sure the sample queues.conf is properly commented.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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e7a20eecb2 |
DTMF attended transfers sometimes fail for no apparent reason.
The loop in feature_request_and_dial() can exit when Party C has answered without processing an AST_CONTROL_ANSWER. Also sometimes an AST_CONTROL_ANSWER never happens even though Party C has answered. Don't hangup Party C if he is up or we receive an AST_CONTROL_ANSWER. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@304005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Terry Wilson
|
9f262f2772 |
Guard against retransmitting BYEs indefinitely
In the case of an attended transfer (A calls B, A atxfers to C) where A becomes unreachable before replying to Asterisk's BYE, Asterisk can sometimes retransmit the BYE indefinitely. This is because __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out, it will not ever be marked as ALREADYGONE, so when __sip_autodestruct is called again, we end up starting the cycle over. This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt in the case of a BYE that has timed out. This should prevent Asterisk from trying to transmit new BYE messages in the future. Review: https://reviewboard.asterisk.org/r/1077/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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c759dd9c8f |
Sending out unnecessary PROCEEDING messages breaks overlap dialing.
Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing through Asterisk. There is not enough information available at this point to know if dialing is complete. The ast_exists_extension(), ast_matchmore_extension(), and ast_canmatch_extension() calls are not adequate to detect a dial through extension pattern of "_9!". Workaround is to use the dialplan Proceeding() application early in non-dial through extensions. * Effectively revert issue #16789. * Allow outgoing overlap dialing to hear dialtone and other early media. A PROGRESS "inband-information is now available" message is now sent after the SETUP_ACKNOWLEDGE message for non-digital calls. An AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE messages for non-digital calls. * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was inconsistent with the cause codes. * Added better protection from sending out of sequence messages by combining several flags into a single enum value representing call progress level. * Added diagnostic messages for deferred overlap digits handling corner cases. (closes issue #17085) Reported by: shawkris (closes issue #18509) Reported by: wimpy Patches: issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664) Expanded upon issue18509_early_media_v1.8_v3.patch to include analog and SS7 because of backporting requirements. Tested by: wimpy, rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303765 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Richard Mudgett
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b3903adbb9 |
Backport the Proceeding application.
Added in trunk -r129399. Enable the workaround for issue #17085 and #18509. (issue #17085) (issue #18509) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303747 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jeff Peeler
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b30fe5496f |
Fix voicemail sequencing for file based storage.
A previous change was made to account for when the number of voicemail messages exceeds the max limit to be handled properly, but it caused gaps in the messages to not be properly handled. This has now been resolved. In later non 1.4 branches, it appears that resequencing wasn't even occurring due from what appears and accidental code removal. (closes issue #18498) Reported by: JJCinAZ Patches: bug18498v2.patch uploaded by jpeeler (license 325) (closes issue #18486) Reported by: bluefox Patches: bug18486.patch uploaded by jpeeler (license 325) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303676 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
Russell Bryant
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a4c9384154 |
Fix a couple of mistakes in the backport of this patch to 1.4.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303552 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Russell Bryant
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58a666ac5a |
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped working properly. This issue includes a patch that resolves the issue by removing a call to ast_check_hangup() from app_meetme.c. I left that in my patch, as it doesn't need to be there. However, the rest of the patch fixes this problem with or without the change to app_meetme. The key difference between what happens before and after this patch is the effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(), ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme sees this which causes it to exit as intended. Checking ast_check_hangup() caused app_meetme to exit earlier in the process, and the target of the redirect saw the condition where ast_read() returned NULL. Removing ast_check_hangup() works around the issue in app_meetme, but doesn't solve the issue if another application did the same thing. There are also other edge cases where if an application finishes at the same time that a redirect happens, the target of the redirect will think that the channel hung up. So, I made some changes in pbx.c to resolve it at a deeper level. There are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to abort the hangup process. My patch extends this to remove the END_OF_Q frame from the channel's read queue, making the "abort hangup" more complete. This same technique was used in every place where a softhangup flag was cleared. (closes issue #18585) Reported by: oej Tested by: oej, wedhorn, russell Review: https://reviewboard.asterisk.org/r/1082/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303546 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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Jason Parker
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518b09e786 |
Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to users.conf configuration. This was surprising users, and potentially setting non-sane "defaults". ASTNOW-125 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@303284 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |