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master
20
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${ noResults }
19699 Commits (cef5e9d4340f94de8801aaecfb004a080df8a68f)
| Author | SHA1 | Message | Date |
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368f8fc4cd |
Merged revisions 307623 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307623 | rmudgett | 2011-02-10 18:29:17 -0600 (Thu, 10 Feb 2011) | 13 lines 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/1.6.2@307624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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daf3853e6c |
Merged revisions 307534 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r307534 | qwell | 2011-02-10 16:33:09 -0600 (Thu, 10 Feb 2011) | 8 lines 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'm reverting here. (issue #18776) Reported by: alecdavis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@307535 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b8e05d1f96 |
Disable color during running test
(closes issue #18776) Reported by: alecdavis Patches: ast_deb_init.diff uploaded by lathama (license 1028) Tested by: andrel, lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@307316 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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36d6bd2396 |
Make sure to set parking dial context for non-default parking lots.
Since parking_con_dial isn't settable, set all parking lots to "park-dial". (closes issue #17946) Reported by: bluecrow76 Patches: asterisk-1.8.0-beta4-multipark-fixes-2010SEP02.diff uploaded by bluecrow76 (license 270) modified by me git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@307227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5fd699dad0 |
Merged revisions 306972 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306972 | twilson | 2011-02-08 12:05:13 -0800 (Tue, 08 Feb 2011) | 2 lines Fix comparison for REFER Replaces tags with pedantic=yes ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@306973 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d5fef6ab68 |
Merged revisions 306965 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) | 1 line fix this line again ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@306966 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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06bdfcedd9 |
Merged revisions 306960 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines 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/1.6.2@306961 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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de3f1d266b |
Merged revisions 306864 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) | 1 line make this safer and fully correct, pointed out by Steve Davis ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@306865 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4cefaad7a0 |
Merged revisions 306672 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306672 | twilson | 2011-02-07 14:35:20 -0800 (Mon, 07 Feb 2011) | 10 lines 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/1.6.2@306673 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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bf7aea5263 |
Merged revisions 306617 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306617 | twilson | 2011-02-07 13:51:43 -0800 (Mon, 07 Feb 2011) | 10 lines 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/1.6.2@306618 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3d510c9a32 |
Don't fallthrough to 'unknown' in the 'ringing' case.
This could cause improper exits from the queue. (closes issue #18499) Reported by: zaltar Patches: app_queue.patch uploaded by zaltar (license 1148) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@306346 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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56ec0fc138 |
Merged revisions 306119 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r306119 | twilson | 2011-02-03 12:36:34 -0800 (Thu, 03 Feb 2011) | 9 lines 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/1.6.2@306126 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b620129580 |
Set exception on channel in parking thread when POLLPRI event detected.
This is done just to make the code be equivalent to the old select code. As noted in 303106 the same issue was already fixed in this branch, but the exception was not set on the channel in the case of POLLPRI. The reason that this did not cause a problem here is because in 122923 the check in __ast_read to check the exception flag was removed. (related to #18637) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@306123 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d21eda6d47 |
res_phoneprov add snom 300, 320, 360, 370, 820, 821, 870 support
(issue #18713) Reported by: lathama Patches: snom_dir.diff uploaded by lathama (license 1028) Tested by: lathama git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305985 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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050a3c39ec |
Merged revisions 305888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines 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(). * Fixed AMI SendText action ast_sendtext() return value check. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305889 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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54dcf83e13 |
Replace link to old doc with new wiki page.
Link to https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305752 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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32447fcfb4 |
SIP Configuration Documentation
sip show settings reports qualifyfreq in milliseconds. sip.conf configures qualifyfreg in seconds. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305648 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7893d91a88 |
Merged revisions 305471 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305471 | qwell | 2011-02-01 11:00:55 -0600 (Tue, 01 Feb 2011) | 9 lines 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/1.6.2@305472 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5f6b31daae |
Merged revisions 305341 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305341 | rmudgett | 2011-01-31 17:45:58 -0600 (Mon, 31 Jan 2011) | 7 lines 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/1.6.2@305342 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1dbde4531e |
Merged revisions 305252 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines 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/1.6.2@305253 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ee372e6cf1 |
Merged revisions 305129 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r305129 | qwell | 2011-01-31 14:56:25 -0600 (Mon, 31 Jan 2011) | 2 lines 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/1.6.2@305130 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f2afe6299a |
Asterisk HTTP response Content-type
Address content type for BSD and other platforms (closes issue #18456) Reported by: alexo Patches: asterisk18_http.patch uploaded by alexo (license 1175) Tested by: alexo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@305082 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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01ad35cc4a |
Merged revisions 304952 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31 Jan 2011) | 2 lines Fix compilation when ODBC_STORAGE is defined. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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54539cc9d9 |
Plug some memory leaks in the LDAP realtime driver.
(closes issue #18435) Reported by: zaltar Patches: res_config_ldap.patch uploaded by zaltar (license 1148) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304865 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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da82320fdd |
If we fail to allocate our announcement objects, make sure we don't leak objects.
The majority of this patch was committed already in r304726 and r304729. (issue #18225) Reported by: kenji (issue #18444) Reported by: junky (closes issue #18343) Reported by: kobaz Patches: meetme-refs.diff uploaded by kobaz (license 834) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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1b79e47b27 |
When we pass the S() or L() options to MeetMe, make sure that we honor C as well.
Without this patch, if the user was kicked from the conference via the S() or L() mechanism, we would just hang up on them even if we also passed C (continue in dialplan when kicked). With this patch we honor the C flag in those cases. (closes issue #17317) Reported by: var git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4ed3d869d9 |
Make sure that we unref the correct object when ejecting the most recent caller.
Currently, when we kick the last user to enter, we decrement our own reference count which results in a crash when we kick another user or when we exit the conference ourselves. This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in 1.6.2. (closes issue #18225) Reported by: kenji Patches: issue18225.patch uploaded by seanbright (license 71) Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304729 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ab0f37345e |
Fix user reference leak in MeetMe.
We were unlinking the user from the conferences user container, but not decrementing the reference count of the user as well, resulting in a leak. (closes issue #18444) Reported by: junky Tested by: seanbright git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304726 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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dc2a91d357 |
Revert part of the previous commit that snuck in.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b1cf1879bc |
Don't leak references if we can't create a pseudo channel for mixing in MeetMe.
If there was a problem allocating a pseudo channel when building our meetme, we weren't destroying our user container or destroying the mutexes that we created. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304659 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c6985c71da |
Merged revisions 304464 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) | 9 lines 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/1.6.2@304465 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7a339ed6ac |
Merged revisions 304460 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304460 | qwell | 2011-01-27 10:47:03 -0600 (Thu, 27 Jan 2011) | 1 line 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/1.6.2@304461 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7fde40240f |
Change delimiter used internally for GOTO_ON_BLINDXFR to commas to match 76703.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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77919e528e |
Merged revisions 304242 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304242 | mmichelson | 2011-01-26 14:38:37 -0600 (Wed, 26 Jan 2011) | 3 lines Get rid of unused 'verbose' field in ast_udptl ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304250 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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051e5a9edf |
Merged revisions 304247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304247 | mnicholson | 2011-01-26 15:00:15 -0600 (Wed, 26 Jan 2011) | 2 lines 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/1.6.2@304249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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afaf018322 |
Merged revisions 304241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304241 | mnicholson | 2011-01-26 14:38:22 -0600 (Wed, 26 Jan 2011) | 6 lines 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/1.6.2@304244 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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632b2a4417 |
Merged revisions 304159 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304159 | seanbright | 2011-01-26 15:18:29 -0500 (Wed, 26 Jan 2011) | 1 line Make sure the sample queues.conf is properly commented. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304181 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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48e5b9d57c |
Merged revisions 304148 from
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier .......... r304148 | rmudgett | 2011-01-26 13:23:46 -0600 (Wed, 26 Jan 2011) | 2 lines Update documentation for DAHDISendCallreroutingFacility() application. .......... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304149 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3ef4c6820e |
Per the man page, setvbuf() must be called before any other operation on an open file.
We use setvbuf() to associate a buffer with a stream, but we have already written to the open file. This works (by chance) on Linux, but fails on other platforms, such as OpenSolaris. (closes issue #16610) Reported by: bklang Patches: setvbuf.patch uploaded by crjw (license 963) Tested by: bklang, asgaroth, efutch git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@304096 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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418ce98827 |
Merged revisions 304005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) | 8 lines 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/1.6.2@304006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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11110b1ead |
Merged revisions 303906 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines 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/1.6.2@303960 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0585c53d3f |
Fix "sip show user <tab>", so that it actually shows results, instead of just completing the last entry.
(closes issue #16675) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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569f4c6604 |
Merged revisions 303765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines 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/1.6.2@303769 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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5369a22c18 |
Merged revisions 303676 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines 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/1.6.2@303677 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines 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/1.6.2@303548 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 303284 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines 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/1.6.2@303285 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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01dfc39eb4 |
Fix changes to L() flag in Dial().
Tony Mountifield pointed out an error I had in my patch. I was a bit too aggressive on changing 'seconds' to 'milliseconds'. So I decided to do some additioanl testing and have no changed just the appropriate lines. One line says milliseconds, and the other says seconds. Probably should change this to be either just seconds or milliseconds, but I've spent too much time on this already :) (issue #18264) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303273 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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main/features: Use POLLPRI when waiting for events on parked channels.
This change resolves a regression in the 1.6.2 when converting from select to poll. The DAHDI timers use POLLPRI to indicate that the timer fired, but features was not waiting for that flag. The result was no audio for MOH when a call was parked and res_timing_dahdi was in use. This patch is slightly modified from the one on the mantis issue. It does not set an exception on the channel if the POLLPRI flag is set. (closes issue #18262) Reported by: francesco_r Patches: patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029) Tested by: francesco_r, rfrantik, one47 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303106 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
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Merged revisions 303007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@303008 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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9aee0b5e0b |
Resolve a compiler warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |