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${ noResults }
4904 Commits (f44ef30a3ed308ecee7de7c143abd8ddb78ce39e)
Author | SHA1 | Message | Date |
---|---|---|---|
|
f44ef30a3e |
Remove dependency on res_features from some channel drivers. It is now part of the core and no longer exists as a module.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100112 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
de9e315146 |
fix flag bit definitions to make code from issue #11049 actually work; along the way, clarify comments and add some dummy flag definitions for other multi-bit flags to hopefully stop this from happening in the future
(closes issue #11049) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100057 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
5fb7250511 |
Merged revisions 99978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7 lines Second attempt. Don't change invitestate when receiving 18x messages in CANCEL state. (issue #11736) Reported by: MVF Patch by oej. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
02e2718e94 |
Merged revisions 99977 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9 lines Make sure we don't cancel destruction on calls in CANCEL state, even if we get 183 while waiting for answer on our CANCEL. (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by oej (license 306) Tested by: MVF ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99979 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
|
0b2239deae |
since echo canceler parameters in Zaptel are now signed integers, allow them during parsing
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e19b83a0a3 |
Progress messages don't work
(closes issue #10497) Reported by: pj Patches: h323-announces-r99483.diff uploaded by sergee (license 138) Tested by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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16315ae2f3 |
- Add a few comments to sip_xmit
- Make sure that we are aware of a pending INVITE even if we're using TCP git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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e956900bc5 |
Merged revisions 99652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4 lines Thanks to Russell's education I realize that BUFSIZ has changed since I learned the C language over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old head to avoid too heavy memory allocations on some systems. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99653 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b8aa3248ec |
Add a generic function to set the bridged call PVT unique id string
as a channel variable BRIDGEPVTCALLID This is important for call tracing in log files and CDRs, so that the SIP callID can be traced along servers. The CHANNEL dialplan function won't work here, since the outbound channel is gone when we need the Call-ID. Other channel drivers may now implement the same function :-), but this patch only supports chan_sip.so. Inspired by (issue #11816) Reported by: ctooley Patch by oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99644 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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949bb30d03 |
Merged revisions 99594 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3 lines Add more dependencies on chan_local and add a note to the description of chan_local so that people don't disable it in menuselect just to clean up. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99596 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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14657e25ea |
Point out a bug in some debug counter handling
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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f4fe6744cc |
Add authentication options to the SIP dialstring.
Documentation follows separately (issue #11587) Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by sobomax (license 359) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99521 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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865bb84432 |
Doxygen updates
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99464 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c48b43e2d8 |
Merged revisions 99426 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21 Jan 2008) | 12 lines Fixing an issue wherein monitoring local channels was not possible. During a channel masquerade, the monitors on the two channels involved are swapped. In 99% of the cases this results in the desired effect. However, if monitoring a local channel, this caused the monitor which was on the local channel to get moved onto a channel which is immediately hung up after the masquerade has completed. By swapping the monitors prior to the masquerade, we avoid the problem by tricking the masquerade into placing the monitor back onto the channel where we want it. During the investigation of the issue, the channel's monitor was the only thing that was swapped in such a manner which did not make sense to have done. All other variable swapping made sense. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99427 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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97fb8a75d4 |
Fix distinctive ring detection.
Reported by: milazzo Patches: drings.diff uploaded by milazzo (license 383) Closes issue #11799 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8f19d8523b |
Remove compiler warning for uninitialized variable
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99384 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fcaeb2d722 |
Doxygen updates.
The TCP/TLS code was committed without any doxygen obviously. Tss tss. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99383 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8296d2d29d |
Updating doxygen
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99382 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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965c454543 |
Merged revisions 99301 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4 lines Bump the buffer size for Via headers up to 512. There are some exceptionally large Via headers out there. (closes issue #11783) Reported by: ofirroval ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99302 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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aeb3048676 |
Change over to using ast_debug so these debug messages don't always show up.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99265 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3a95a07e69 |
Add a "console active" CLI command, which lets you find out which console device
is currently active for the Asterisk CLI, or to set it. Also, knock multiple device support off of the to-do list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99248 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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f20450ea03 |
Merge changes from team/russell/console_devices
- Add support for multiple devices. All devices are configured in console.conf. - Add "console list devices" CLI command to show configured devices. Also, changed the old "list devices" to be "list available", which queries PortAudio for all audio devices that are available for use. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b995c78c31 |
Merge changes from team/group/sip-tcptls
This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99085 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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89778a05c1 |
Make the output of "console list devices" a bit prettier.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99011 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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30de4fc350 |
List which devices are inputs and outputs in "console list devices"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99009 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2a91da6613 |
Merged revisions 99004 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines Have IAX2 optimize the codec translation path just like chan_sip does it. If the caller's codec is in our codec list, move it to the top to avoid transcoding. (closes issue #10500) Reported by: stevedavies Patches: iax-prefer-current-codec.patch uploaded by stevedavies (license 184) iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184) Tested by: stevedavies, pj, sheldonh ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8dc5e09ccb |
Add several busy detection related defines to menuselect.
Allow better busy detect debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches: busydetect_enhancement.patch uploaded by agx (license 298) busydetect-r94975.diff uploaded by sergee (license 138) Additional changes/cleanup by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98998 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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cd4cc27c93 |
major reliability and performance improvement in VWMI monitoring for FXO ports (code by markster, me and dbailey)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98990 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6aaa992301 |
Merge the changes from issue #10665 from the team/group/sip_session_timers branch.
This set of changes introduces SIP session timers support (RFC 4028). In short, this prevents stuck SIP sessions that were not properly torn down due to network or endpoint failures during an established SIP session. To quote some of the documentation supplied with the patch: "The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE request at a negotiated interval. If a session refresh fails then all the entities that support Session- Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path that do not support Session-Timers)." (closes issue #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by rjain (license 226) chan_sip.c.diff uploaded by rjain (license 226) sip.conf.sample.diff uploaded by rjain (license 226) proc_422_rsp_comment.diff uploaded by rjain (license 226) chan_sip.c.cache.diff uploaded by rjain (license 226) chan_sip.memalloc uploaded by rjain (license 226) chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches tracked in team/group/sip_session_timers, with some additional fixes by russell and oej. Tested by: jtodd, rjain, loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a6da07a515 |
Merged revisions 98966 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6 lines Add missing NULLs at end of two ast_load_realtimes. (closes issue #11769) Reported by: tequ Patches: chaniax.patch uploaded by dimas (license 88) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98967 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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343949b741 |
Merged revisions 98964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 Jan 2008) | 10 lines Fix a deadlock in chan_local in local_hangup. There was contention because the local_pvt was held and it was attempting to lock a channel, which is the incorrect locking order. (closes issue #11730) Reported by: UDI-Doug Patches: 11730.patch uploaded by putnopvut (license 60) Tested by: UDI-Doug ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98965 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4082bed03a |
Merged revisions 98955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 lines Don't drop the old record route information when dealing with packets related to a reinvite. (closes issue #11545) Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by kebl0155 (license 356) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98956 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1faba2a90c |
Remove DNS lookup from sip_devicestate. This seems to come from way back when and I can't think of a reason for it being here, plus it could cause needless DNS lookups.
(closes issue #10983) Reported by: jtodd git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2cdf636c0f |
Merged revisions 98946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | 11 lines Change a buffer in check_auth() to be a thread local dynamically allocated buffer, instead of a massive buffer on the stack. This fixes a crash reported by Qwell due to running out of stack space when building with LOW_MEMORY defined. On a very related note, the usage of BUFSIZ in various places in chan_sip is arbitrary and careless. BUFSIZ is a system specific define. On my machine, it is 8192, but by definition (according to google) could be as small as 256. So, this buffer in check_auth was 16 kB. We don't even support SIP messages larger than 4 kB! Further usage of this define should be avoided, unless it is used in the proper context. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98948 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4fb04cb58a |
Merged revisions 98943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines Commit a fix for some memory access errors pointed out by the valgrind2.txt output on issue #11698. The issue here is that it is possible for an instance of a translator to get destroyed while the frame allocated as a part of the translator is still being processed. Specifically, this is possible anywhere between a call to ast_read() and ast_frame_free(), which is _a lot_ of places in the code. The reason this happens is that the channel might get masqueraded during this time. During a masquerade, existing translation paths get destroyed. So, this patch fixes the issue in an API and ABI compatible way. (This one is for you, paravoid!) It changes an int in ast_frame to be used as flag bits. The 1 bit is still used to indicate that the frame contains timing information. Also, a second flag has been added to indicate that the frame came from a translator. When a frame with this flag gets released and has this flag, a function is called in translate.c to let it know that this frame is doing being processed. At this point, the flag gets cleared. Also, if the translator was requested to be destroyed while its internal frame still had this flag set, its destruction has been deffered until it finds out that the frame is no longer being processed. Admittedly, this feels like a hack. But, it does fix the issue, and I was not able to think of a better solution ... ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9a76fbf9c2 |
Merged revisions 98934 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4 lines Based on the boundary found move over the correct amount. (closes issue #11750) Reported by: tasker ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98935 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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698ad33d7b |
Merged revisions 98894 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4 lines Accept "; boundary=" not just ";boundary=" in the multipart mixed content type. (closes issue #11750) Reported by: tasker ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98895 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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524aed3768 |
Add proper call forwarding (all and busy) support for chan_skinny.
Note: NoAnswer support is currently not implemented, as it would take a significant amount of work to figure out how to do correctly. Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98776 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d357378397 |
Fix for potential crash with vmexten
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4b0a63ffa2 |
Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always gets generated.
(closes issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded by tzafrir (modified by me) (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98488 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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138799091c |
Add 'auto' signalling mode for Zaptel channels.
(closes issue #11690) Reported by: tzafrir Patches: signaling_to_signalling.diff uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir (license 46) zap_no_default_sig.diff uploaded by tzafrir (license 46) zap_signal_auto.diff uploaded by tzafrir (license 46) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98436 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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911fbb5df9 |
Merged revisions 98164 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) | 2 lines Back out changes from revision 97077, since it wasn't perfect ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98193 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9603b5f598 |
Ascom phones send Flash events as SIP INFO using '!' as the 'digit'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98124 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5c2beee6c3 |
Add a new global and per-peer option to chan_sip, qualifyfreq, which allows you
to set the qualify frequency. (closes issue #11597) Reported by: wilder Patches: qualifyfreq5.patch uploaded by wilder (license 362) -- with some mods by me git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98027 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c88f243d8d |
Merged revisions 97973 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines 1) When we get a translated frame out, clone it, because if the translator pvt is freed before we use the frame, bad things happen. 2) Getting a failure from ast_sched_delete means that the schedule ID is currently running. Don't just ignore it. (Closes issue #11698) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97978 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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71ad27409c |
The fixes in this commit are mainly to allow compiling of trunk with --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly, the version.c needs to be in the OBJS line; asterisk.h was chosen to have the prototypes for ast_get_version, ast_get_version_num; and the ASTERISK_FILE_VERSION macro needs to be used after including asterisk.h in a few files. I hope I did the right thing. If not, let me know.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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857e3412f4 |
Several manager changes:
1) Add the Dialplan class, for NewExten and VarSet events, which should cut down on the volume of traffic in the Call class. 2) Permit some commands to be run from multiple classes, such as allowing DBGet to be run from either the System or the Reporting class. 3) Heavily document each class in the sample config, as there were several that made no sense to be in the write= line, and two that made no sense to be in the read= line (since they controlled no permissions there). (Closes issue #10386) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97651 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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84d4c3d801 |
remove a wrong 'const'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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cdf6974dea |
Two changes:
- support scrolling of message window; - simplify the code for creating a message window, and try it using a second one in the top of the keypad (where we echo the dialed number). The 'skin' that supports these two windows will be committed separately. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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55240a4e35 |
Merged revisions 97489 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines Set the caller id within the gtalk_alloc function. As underlined in issue #10437 by Josh, we need to prevent a possible memory leak. We only set the name part of the caller id, the number part is not relevant when dealing with JIDs. Closes issue #11549. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |