mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
18854 Commits (117d3fd910c0ddda73d5f782bfbac235a9c44a46)
Author | SHA1 | Message | Date |
---|---|---|---|
|
117d3fd910 |
Add the programs in utils/ to menuselect.
Nothing in utils/ is now built by default except for astcanary. Review: https://reviewboard.asterisk.org/r/353/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225440 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
d9f72c1893 |
Permit storage of voicemail secrets in a separate file, located within the spool directory.
(closes issue #14276) Reported by: klaus3000 Patches: app_voicemail.c-svn-trunk-r214898.txt uploaded by klaus3000 (license 65) Tested by: jamesgolovich git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225406 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
be796e703a |
Fix a refcount error introduced by yesterday's OBJ_MULTIPLE commit.
When an object is being unlinked from its container *and* being returned to the caller, we do not want to decrement the reference count after unlinking it from the container, as the reference that the container held is what we are returning to the caller... and if it was the only remaining reference to the object, that could result in the object being destroyed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225405 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
496282194c |
Merged revisions 225105 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines Fix documentation for ast_softhangup() and correct the misuse thereof. (closes issue #16103) Reported by: majorbloodnok ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225360 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
1174a61612 |
Add support for calling and called subaddress. Partial support for COLP subaddress.
The Telecom Specs in NZ suggests that SUB ADDRESS is always on, so doing "desk to desk" between offices each with an asterisk box over the ISDN should then be possible, without a whole load of DDI numbers required. (closes issue #15604) Reported by: alecdavis Patches: asterisk_subaddr_trunk.diff11.txt uploaded by alecdavis (license 585) Some minor modificatons were made. Tested by: alecdavis, rmudgett Review: https://reviewboard.asterisk.org/r/405/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
3acfd4933c |
Merged revisions 225243 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225243 | dvossel | 2009-10-21 15:58:08 -0500 (Wed, 21 Oct 2009) | 13 lines IAX2: VNAK loop caused by signaling frames with no destination call number It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225307 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
87ff40d3f3 |
Add 'mohsuggest' configuration option to 'sip show peer' CLI command and
SIPShowPeer AMI action. (closes issue #15990) Reported by: _brent_ Patches: sip_peer_info_mohsuggest-r3.patch uploaded by brent (license 388) Review: https://reviewboard.asterisk.org/r/381/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225245 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
cdd1f9e296 |
Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it.
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the case where multiple results need to be returned; OBJ_NODATA mode already was supported). In addition, it converts ast_channel_iterators (only the targeted versions, not the ones that iterate over all channels) to use this method. During this work, I removed the 'ao2_flags' arguments to the ast_channel_iterator constructor functions; there were no uses of that argument yet, there is only one possible flag to pass, and it made the iterators less 'opaque'. If at some point in the future someone really needs an ast_channel_iterator that does not lock the container, we can provide constructor(s) for that purpose. Review: https://reviewboard.asterisk.org/r/379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
1ebf7767d0 |
Merged revisions 225171 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225171 | russell | 2009-10-21 11:44:49 -0500 (Wed, 21 Oct 2009) | 2 lines Revert 225169, as this doesn't account for the possibility of a list of frames. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225172 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
9fbb9d0b6c |
Merged revisions 225169 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225169 | russell | 2009-10-21 11:39:20 -0500 (Wed, 21 Oct 2009) | 2 lines Isolate the frame returned from ast_translate(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225170 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
d01cf285dd |
Blocked revisions 225103 via svnmerge
........ r225103 | tilghman | 2009-10-21 10:45:54 -0500 (Wed, 21 Oct 2009) | 2 lines Suffix is not needed for a match ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225104 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
0776bcff64 |
Apparently, I don't need to specify the ".so" suffix to get a match
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225102 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
|
01ab66275a |
Add support for specifying the IP address to use for media streams in sip.conf
This is the second commit for this and documents the text stream using the configured IP address and fixes a bug in the original patch where the UDPTL stream would also use the different IP address. (closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225089 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a2f809c127 |
Turn on DENOISE filter for all conference participants.
(Fixes SWP-238) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225048 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a31eb5bb35 |
Revert media_address commit, I'm going to roll a fix to the SDP generation in the next version.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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984d6500ce |
Merged revisions 225032 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r225032 | dvossel | 2009-10-21 09:37:04 -0500 (Wed, 21 Oct 2009) | 20 lines IAX/SIP shrinkcallerid option The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225033 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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28d0ec5421 |
Add support for specifying the IP address to use for media streams in sip.conf
(closes issue #14729) Reported by: _brent_ Patches: media_address.patch uploaded by brent (license 388) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225003 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cd10bd931a |
Merged revisions 224931 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224931 | russell | 2009-10-20 21:59:54 -0500 (Tue, 20 Oct 2009) | 5 lines Isolate frames returned from a DSP instance or codec translator. The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224932 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c5b8e9af7c |
Make PRI_SUBCMD_xxx handling subaddress friendly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224930 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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77031501a5 |
Merged revisions 224855 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224855 | tilghman | 2009-10-20 17:07:11 -0500 (Tue, 20 Oct 2009) | 5 lines Pay attention to the return value of the manipulate function. While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b518c7b9a7 |
Merged revisions 224773 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224773 | file | 2009-10-20 14:46:37 -0300 (Tue, 20 Oct 2009) | 5 lines Add support for relaying early media in the features attended transfer option. (closes issue #14828) Reported by: licedey ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224774 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e90ff8de4c |
Added information to CHANGES about the dynamic range compression feature added to dahdi.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224738 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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092a118d89 |
Merged revisions 224670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines Correct timestamp calculations when RTP sample rates over 8kHz are used. While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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26638d3a55 |
Add dynamic range compression support for analog channels.
(closes issue AST-29) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b7a50aeddc |
Merged revisions 224565 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224565 | file | 2009-10-19 16:47:50 -0300 (Mon, 19 Oct 2009) | 5 lines Do not attempt early media bridging (ie: direct RTP setup) if options are enabled that should prevent it. (closes issue #14763) Reported by: cupotka ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224567 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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39e730b4e2 |
Remove useless debugging message.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224562 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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2b82d051ca |
Remove a completed project and add another
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224527 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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fd9f9ab01e |
Add a callback to sig_pri which is called when sig_pri is going to queue a control frame on a channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224491 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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8f9edddc27 |
Allow ODBC storage to be queried with multiple mailboxes, and remove multiple goto's.
This corrects an issue reported on the -users list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224448 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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97bf6e881a |
Clarify that "forcecommit" is NOT an alias for "autocommit", but instead controls the default disposition of uncommitted transactions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c80715706e |
Remove unnecessary typedef
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224403 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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03db5ef0e5 |
fix typo, sorry
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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53a95d9c84 |
Merged revisions 224330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224330 | jpeeler | 2009-10-16 20:32:47 -0500 (Fri, 16 Oct 2009) | 13 lines Fix stale caller id data from being reported in AMI NewChannel event The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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64a32b3ad0 |
Merged revisions 224260 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r224260 | rmudgett | 2009-10-16 15:25:23 -0500 (Fri, 16 Oct 2009) | 18 lines Never released PRI channels when using Busy() or Congestion() dialplan apps. When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (issue #14292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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c74a2d0b45 |
Create an API for adding an optional time unit onto the ends of time periods.
Two examples of its use are included, and the usage could be expanded in some cases into certain configuration options where time periods are specified. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224225 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a39f3a7521 |
Readd removed ability to allow listening to one side of the call in app_chanspy
(Option o) (closes issue #15675) Reported by: john8675309 Patches: issue15675patchtrunk.txt uploaded by dbrooks (license 790) Tested by: jgutierrez on users list: http://lists.digium.com/pipermail/asterisk-users/2009-October/239155.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224178 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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fb1433f43f |
chan_dahdi.conf.sample changes for DTMF CID detect
Explains new options for detecting DTMF CID on fxo lines (issue #9096) Reported by: fleed Patches: chan_dahid_sample_config.patch uploaded by sum (license 766) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224144 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5af2438403 |
Properly handle PUT requests for CALENDAR_WRITE()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224109 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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cb74681b6d |
Add missing 'getnum' field
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224074 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e3f473f4f3 |
Allow for adding message body to the SIP NOTIFY message
Ability has been added to both manager command SIPnotify as well as console command sip notify. Message body is stored in the "Content" variable. An example is present in sip_notify.conf. (closes issue #13926) Reported by: jthurman Patches: sip-notify-svn189463.diff uploaded by gareth (license 208) Tested by: gareth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224035 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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5ea7d4a291 |
use Calendar: instead of Calendar/ for devstate
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223992 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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51f34d24e4 |
Fix some doxygen format problems and trim trailing whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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76831d0c78 |
Fix compiler warning.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223911 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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1f9f1562ba |
Revert inadvertant code commit to app_originate
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223875 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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a8034cd770 |
Fix handling of notification calls w/ the dialing api
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223874 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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832be82dfb |
Merged revisions 223804 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223804 | jpeeler | 2009-10-12 18:12:50 -0500 (Mon, 12 Oct 2009) | 8 lines Ensure ringing continues for branched calls after progress is received While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223832 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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b14857f49d |
Clarifies trunkmaxsize, trunkfreq, and trunkmtu iax2 options
SWP-151 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223756 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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7da0353ebd |
Recorded merge of revisions 223692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r223692 | kpfleming | 2009-10-12 10:30:40 -0500 (Mon, 12 Oct 2009) | 13 lines Remove automatic switching from T.38 to voice mode in chan_sip. chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover. (issue #16025) Reported by: jamicque ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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e197f85b8c |
Remove automatic switching from T.38 to voice mode in chan_sip.
chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover, and changes app_fax to explicitly switch off T.38 mode when the FAX transmission process ends. (closes issue #16025) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223652 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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9e1598b762 |
Check the proper page for the SENDRPID flag.
If a pending reinvite were sent, we might not properly send connected party info since we were checking the wrong flag. This was a rare occurrence, but could still happen nevertheless. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@223617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |