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${ noResults }
17385 Commits (1448b5db6a5570f9129d1599bb224e7309d5b31d)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Kevin P. Fleming
|
1448b5db6a |
correct warning message to not refer specifically to DAHDI
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175250 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Jeff Peeler
|
66e88633a5 |
Merged revisions 175187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Vossel
|
53f3ab973e |
Setting key rotation to be off by default
Key rotation breaks compatibility between (trunk/1.6.1) and (1.2/1.4/1.6.0). As a follow up to this, I am investigating possible ways to allow key rotation to be on by default and not affect the other branches, but for now it must be turned off. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175127 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
12f02a8c11 |
Merged revisions 175124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175125 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
3a9d79f056 |
Make lock information for ao2_trylock be more useful and gnarly
Core show locks information involving an ao2_trylock did not show the function that called ao2_trylock, but would instead show ao2_trylock as the source of the lock. This is not useful when trying to debug locking issues. One bizarre note is that this logic is already in 1.4 but somehow did not get merged to trunk or the 1.6.X branches. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175121 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Philippe Sultan
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c873d72ca2 |
Issue a warning message if our candidate's IP is the loopback address.
(closes issue #13985) Reported by: jcovert Tested by: phsultan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Philippe Sultan
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7bb5ef8399 |
Merged revisions 175029 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175029 | phsultan | 2009-02-12 11:16:21 +0100 (Thu, 12 Feb 2009) | 12 lines Set the initiator attribute to lowercase in our replies when receiving calls. This attribute contains a JID that identifies the initiator of the GoogleTalk voice session. The GoogleTalk client discards Asterisk's replies if the initiator attribute contains uppercase characters. (closes issue #13984) Reported by: jcovert Patches: chan_gtalk.2.patch uploaded by jcovert (license 551) Tested by: jcovert ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@175058 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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0d5da5f436 |
Fix a bit of odd logic for announcing position. Sync with 1.6.0's logic
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174951 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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34161542e9 |
Fix odd "thank you" sound playing behavior in app_queue.c
If someone has configured the queue to play an position or holdtime announcement, then it is odd and potentially unexpected to hear a "Thank you for your patience" sound when no position or holdtime was actually announced. This fixes the announcement so that the "thanks" sound is only played in the case that a position or holdtime was actually announced. There is a way that the "thank you" sound can be played without a position or holdtime, and that is to set announce-frequency to a value but keep announce-position and announce-holdtime both turned off. (closes issue #14227) Reported by: caspy Patches: 14227_v3.patch uploaded by putnopvut (license 60) Tested by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174948 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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47ebea6a8d |
Fix 'd' option for app_dial and add new option to Answer application
The 'd' option would not work for channel types which use RTP to transport DTMF digits. The only way to allow for this to work was to answer the channel if we saw that this option was enabled. I realized that this may cause issues with CDRs, specifically with giving false dispositions and answer times. I therefore modified ast_answer to take another parameter which would tell if the CDR should be marked answered. I also extended this to the Answer application so that the channel may be answered but not CDRified if desired. I also modified app_dictate and app_waitforsilence to only answer the channel if it is not already up, to help not allow for faulty CDR answer times. All of these changes are going into Asterisk trunk. For 1.6.0 and 1.6.1, however, all the changes except for the change to the Answer application will go in since we do not introduce new features into stable branches (closes issue #14164) Reported by: DennisD Patches: 14164.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/145 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
ff0b908777 |
Blocked revisions 174885 via svnmerge
........ r174885 | tilghman | 2009-02-11 14:54:18 -0600 (Wed, 11 Feb 2009) | 13 lines Restore a behavior that was recently changed, when we fixed issue #13962 and issue #13363 (related to issue #6176). When a hangup occurs during a Macro execution in earlier 1.4, the h extension would execute within the Macro context, whereas it was always supposed to execute only within the main context (where Macro was called). So this fix checks for an "h" extension in the deepest macro context where a hangup occurred; if it exists, that "h" extension executes, otherwise the main context "h" is executed. (closes issue #14122) Reported by: wetwired Patches: 20090210__bug14122.diff.txt uploaded by Corydon76 (license 14) Tested by: andrew ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174886 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
8435535300 |
Tell the device state core a change happened when a channel is freed but not a specific state.
We need to do this because while we know that the freeing of the channel may cause something to become not in use we do not know this for sure. There may be another channel that is still up which would cause it to be in use. (closes issue #13238) Reported by: kowalma Patches: 20090121__bug13238.diff.txt uploaded by Corydon76 (license 14) Tested by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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c9670b24a4 |
Fix potential for stack overflows in app_chanspy.c
When using the 'g' or 'e' options, the stack allocations that were used could cause a stack overflow if a spyer stayed on the line long enough without actually successfully spying on anyone. The problem has been corrected by using static buffers and copying the contents of the appropriate strings into them instead of using functions like alloca or ast_strdupa git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174805 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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e909bfcea9 |
Fix an fd leak that would occur in HTTP AMI sessions
The explanation behind this fix is a bit complicated, and I've already typed it up in the code as a huge comment inside of manager.c, so I'll give the abridged version here. We needed a way to separate action-specific data from session-specific data. Unfortunately, the only way to maintain API compatibility and to not have to change every single manager action was to rename the current mansession structure and wrap it inside a new mansession structure which actually contains action- specific data. (closes issue #14364) Reported by: awk Patches: 14364_better.patch uploaded by putnopvut (license 60) Tested by: putnopvut Review: http://reviewboard.digium.com/r/148/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174764 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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6304c09149 |
Only decrease inringing count if above zero.
(issue #13238) Reported by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174710 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Kevin P. Fleming
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23939e54f3 |
improve slinfactory API to remove implicit sample rate and require explicit sample rate selection by creator of the slinfactory
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174705 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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656110e42b |
Blocked revisions 174644 via svnmerge
........ r174644 | file | 2009-02-10 14:50:50 -0400 (Tue, 10 Feb 2009) | 6 lines Go off hold when we get an empty reinvite telling us to. (closes issue #14448) Reported by: frawd Patches: hold_invite_nosdp.patch uploaded by frawd (license 610) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174645 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Matthew Nicholson
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b7dc074376 |
Merged revisions 174583 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174583 | mnicholson | 2009-02-10 11:52:42 -0600 (Tue, 10 Feb 2009) | 18 lines Improve behavior of jitterbuffer when maxjitterbuffer is set. This change improves the way the jitterbuffer handles maxjitterbuffer and dramatically reduces the number of frames dropped when maxjitterbuffer is exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all new frames were dropped until the jitterbuffer is empty. This change modifies the code to only drop frames until maxjitterbuffer is no longer exceeded. Also, previously when maxjitterbuffer was exceeded, dropped frames were not tracked causing stats for dropped frames to be incorrect, this change also addresses that problem. (closes issue #14044) Patches: bug14044-1.diff uploaded by mnicholson (license 96) Tested by: mnicholson Review: http://reviewboard.digium.com/r/144/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174584 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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8e6780a5b1 |
Set the type for the peer structure to be a peer as the default.
(closes issue #14447) Reported by: triccyx git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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bb327036f1 |
Make the logic for inuse and inringing manipluation match that of 1.4. The old broken logic would reset the values back to 0 during certain scenarios causing the wrong state to be reported.
(closes issue #14399) Reported by: caspy (issue #13238) Reported by: kowalma git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174543 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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d4df1c6a78 |
Fix0ring build
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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51c80f5b40 |
Remove the usage of the KeepAlive app, as it no longer exists.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Steve Murphy
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3084e9ce93 |
This patch removes the use of AST_PBX_KEEPALIVE
from app_rpt.c. (closes issue #14435) Reported by: D_McNaul git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174435 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Steve Murphy
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43462557c0 |
More intptr_t work.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174432 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Steve Murphy
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dbe56b6647 |
Merged revisions 174369 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174369 | murf | 2009-02-09 19:27:40 -0700 (Mon, 09 Feb 2009) | 5 lines This patch solves some compiler complaints in both 32 and 64-bit environments. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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e0b0ae07a3 |
Fix something I messed up in the merge I just did
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174327 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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829b14fc3e |
Fixes issue with hangups not being sent and external process never terminating.
The ignore_hangup, run_dead, and noanswer flags were never initilized to zero causing hangups to never be issued. If the external script expects to be notified of a hangup and never receives one, it runs indefinitely. (closes issue #14251) Reported by: chris-mac Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174325 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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a02ef73b25 |
Merged revisions 174282 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174282 | mmichelson | 2009-02-09 11:11:05 -0600 (Mon, 09 Feb 2009) | 12 lines Don't do an SRV lookup if a port is specified RFC 3263 says to do A record lookups on a hostname if a port has been specified, so that's what we're going to do. See section 4.2. (closes issue #14419) Reported by: klaus3000 Patches: patch_chan_sip_nosrvifport_1.4.23.txt uploaded by klaus3000 (license 65) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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23760c47d3 |
Merged revisions 174218 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174218 | file | 2009-02-09 10:48:21 -0400 (Mon, 09 Feb 2009) | 4 lines Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off. (closes issue #14407) Reported by: mostyn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174219 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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e77b3cea6b |
Merged revisions 174148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174148 | russell | 2009-02-07 10:15:07 -0600 (Sat, 07 Feb 2009) | 2 lines Fix a race condition that could cause a crash. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174149 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Dwayne M. Hubbard
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0024ad62ab |
Merged revisions 174082 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r174082 | dhubbard | 2009-02-06 17:36:03 -0600 (Fri, 06 Feb 2009) | 5 lines check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp() and sip_uri_params_cmp() The reporter didn't actually upload a properly-formed patch, instead a modified chan_sip.c file was uploaded. I created a patch to determine the changes, then modified the suggested changes to create a proper fix. The summary above is a complete description of the changes. (closes issue #13547) Reported by: tecnoxarxa Patches: chan_sip.c.gz uploaded by tecnoxarxa (license 258) Tested by: tecnoxarxa ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174084 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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c15b83e7e5 |
Adds immediate yes/no option to iax.conf
This is very similar to the DAHDI immediate=yes option. When the phone is picked up, instead of giving a dialtone it connects directly to the "s" extension. Changes where implemented in chan_iax2.c to directly connect to the "s" extension in the appropriate context when this option is enabled. Examples explaining its use are added to iax2.conf.sample. CHANGES has been updated as well. (closes issue #14266) Reported by: jcovert Patches: chan_iax2.c.patch-trunk uploaded by jcovert (license 551) iax.conf.sample.patch uploaded by jcovert (license 551) Tested by: jcovert, dvossel Review: http://reviewboard.digium.com/r/143/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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fa389ab17a |
Don't subscribe to a mailbox on pseudo channels. It is futile. This solves an issue where duplicated pseudo channels would cause a crash because the first one would unsubscribe and the next one would also try to unsubscribe the same subscription.
(closes issue #14322) Reported by: amessina git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@174041 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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550f7f1e65 |
Merged revisions 173967-173968 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173967 | file | 2009-02-06 13:14:15 -0400 (Fri, 06 Feb 2009) | 4 lines Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string. (closes issue #14350) Reported by: fhackenberger ........ r173968 | file | 2009-02-06 13:15:01 -0400 (Fri, 06 Feb 2009) | 2 lines Remove a debug message I put in by accident. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173974 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Matthew Nicholson
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647b68ec23 |
Merged revisions 173917 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173917 | mnicholson | 2009-02-06 10:20:23 -0600 (Fri, 06 Feb 2009) | 7 lines Limit the addition of the Contact header in SIP responses according to various SIP RFCs. (closes issue #13602) Reported by: hjourdain Tested by: mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173952 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Joshua Colp
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b2484d7db9 |
Always detach and destroy the whisper and barge audiohooks. Additionally also allow an audiohook to be detached if it has not been attached.
(closes issue #14414) Reported by: bluecrow76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173902 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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5da26420e9 |
Blocked revisions 173900 via svnmerge
........ r173900 | tilghman | 2009-02-06 09:43:32 -0600 (Fri, 06 Feb 2009) | 3 lines Backport OS X fix from trunk (AGAIN, closes issue #14360) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173901 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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0cbada4f96 |
Add a common implementation of a scheduler context with a dedicated thread.
This commit expands the Asterisk scheduler API to include a common implementation of a scheduler context being processed by a dedicated thread. chan_iax2 has been updated to use this new code. Also, as a result, this resolves some race conditions related to the previous chan_iax2 scheduler handling. Related to rev 171452 which resolved the same issues in 1.4. Code from team/russell/sched_thread2 Review: http://reviewboard.digium.com/r/129/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Russell Bryant
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1d46bb0d0f |
Resolve a memory leak that would occur on an invalid channel given to Action: Status
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173848 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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69dff2f5f8 |
Update extensions.conf.sample to be correct.
In trunk, the only necessary change pointed out was that the call to ChanIsAvail uses an option that has been removed. For the 1.6.1 branch, however, it appears that the sample file is badly in need of updating since there are |'s used all over the place there. My tentative plan is just to copy trunk's sample config file to those branches since the info there is most up-to-date and should be correct for use in 1.6.1 Thanks to macli in #asterisk-dev for bringing this up git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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8f9c8f8b40 |
Properly set "seen" and "unseen" flags when moving messages from the new to the old folder when using IMAP for voicemail storage
(closes issue #13905) Reported by: jaroth Patches: foldermove_v2.patch uploaded by jaroth (license 50) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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bf5358dd26 |
Blocked revisions 173770 via svnmerge
........ r173770 | mmichelson | 2009-02-05 17:19:16 -0600 (Thu, 05 Feb 2009) | 20 lines Fix logic regarding when to perform an SRV lookup for outgoing REGISTER requests With this fix, we only will perform an SRV lookup at the following times: * The first time we register with a remote registrar * If we send a REGISTER but do not receive a response * If the sendto() function returns an error While I wrote the patch that fixes this issue, a huge amount of credit is due to Brett Bryant, who wrote the initial patch on which I based this one. (closes issue #12312) Reported by: jrast Patches: 12312.patch uploaded by putnopvut (license 60) Tested by: blitzrage Review: http://reviewboard.digium.com/r/132/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Jeff Peeler
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4ac807845d |
Merged revisions 173696 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173696 | jpeeler | 2009-02-05 14:47:51 -0600 (Thu, 05 Feb 2009) | 12 lines Add new configuration option to make shared IMAP mailboxes function as expected. The new option is "imapvmshareid" which is an ID to tag multiple mailboxes using the same IMAP storage location to function as one mailbox. This allows all messages to be retrieved for any user in the group. The patch alters the 'X-Asterisk-VM-Extension' header that is responsible for matching voicemails for a given user. (closes issue #13673) Reported by: howardwilkinson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173697 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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2e1d9f9a21 |
Merged revisions 173692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173692 | mmichelson | 2009-02-05 14:29:09 -0600 (Thu, 05 Feb 2009) | 12 lines Fix situations where queue members could be autopaused unexpectedly Specifically, this patch prevents us from autopausing members when we receive a busy or congestion frame from them. (closes issue #14376) Reported by: fiddur Patches: 14376.patch uploaded by putnopvut (license 60) Tested by: fiddur ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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4cc1606d27 |
Change the first field, or we don't get the necessary field separation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173657 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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5d3fc1ee3a |
Merged revisions 173592 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173592 | mmichelson | 2009-02-05 12:47:24 -0600 (Thu, 05 Feb 2009) | 3 lines Add some missing cleanup to app_mixmonitor ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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5f83636af4 |
Merged revisions 173559 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173589 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Mark Michelson
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172777bd02 |
Fix some areas where the incorrect interface was passed to ast_device_state
I swear it feels like I already did this once... (closes issue #14359) Reported by: francesco_r git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173507 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Tilghman Lesher
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f931abc61a |
Add XML documentation for the applications and functions in res_jabber
(closes issue #14405) Reported by: snuffy Patches: xml_jabber.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173503 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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David Vossel
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b605bdff0c |
Fixes issue with IAX2 transfer not handing off calls. Reverts changes in 116884
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table. The changes reverted in 116884 caused backwards compatibility issues involving iax2 transfer with 1.6.0, 1.4, and 1.2. (closes issue #13468) Reported by: nicox Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |