mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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877 Commits (3913dcbc40039ca3c1942b7300fc45daeb4970ea)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
05899c19a1 |
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
17 years ago |
|
|
ed1951d895 |
Merge code associated with AST-2009-006
(closes issue #12912) Reported by: rathaus Tested by: tilghman, russell, dvossel, dbrooks git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@216000 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
3300ac5b40 |
Clarify queues.conf comments to specify that variables should be set in the dialplan.
(closes issue #15755) Reported by: trendboy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@213493 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
361c9a99e1 |
Allow for UDPTL to use only even-numbered ports if desired.
There are some VoIP providers out there that will not accept SDP offers with odd numbered UDPTL ports. While it is my personal opinion that these VoIP providers are misinterpreting RFC 2327, it really is not a big deal to play along with their silly little games. Of course, since restricting UDPTL ports to only even numbers reduces the range of available ports by half, so the option to use only even port numbers is off by default. A user can enable the behavior by setting use_even_ports=yes in udptl.conf. (closes issue #15182) Reported by: CGMChris Patches: 15182.patch uploaded by mmichelson (license 60) Tested by: CGMChris git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@209131 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
1e30dcf61c |
Enhance configuration option for overlapdial allowing direction choice
Previously overlap dialing could only be turned on or off for both incoming and outgoing calls. New parameters incoming, outgoing, and both have been added to allow further control. There is no change in default behavior with these new options and allows in band DTMF to be accepted in one direction if required. (closes issue #14471) Reported by: eboscani git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@207092 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
5510a1c74e |
error in iax.conf related IP-based access control
(closes issue #15518) Reported by: pkempgen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@206872 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
5b9004d067 |
Fix some spelling fail.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@194764 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
58b5a85e80 |
Make absolute paths for logger channels work properly
(Note: This is not a new feature, it was previously undocumented and broken.) The Asterisk logger has a feature to support absolute pathnames for logger channels, but the code implementing the feature was broken. This has been fixed, and the absolute path feature is now documented in the sample logger.conf. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@193193 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
6d08bad538 |
Distinguish in a sent email between simple sends and forwards.
(closes issue #11678) Reported by: jamessan Patches: 20090330__bug11678.diff.txt uploaded by tilghman (license 14) Tested by: tilghman, lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186415 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
09490bb688 |
Fix instructions in one-step parking comment to make more sense.
Changed a capital K to a lowercase k. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
24fa699663 |
Merged revisions 186056 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r186056 | tilghman | 2009-04-02 12:02:18 -0500 (Thu, 02 Apr 2009) | 2 lines Fix for AST-2009-003 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@186059 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
63ca43071e |
Update the channel allocation method documentation.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@185121 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
|
cf7131dd6a |
Backport state interface changes to app_queue from trunk.
After several issues raised on the Asterisk bugtracker against the 1.4 branch were determined to be fixable with the state interface change available in the 1.6.X series, it finally came time to just suck it up and backport the change. For a detailed explanation of what this change entails, the original trunk commit for this feature may be found here: http://svn.digium.com/view/asterisk?view=revision&revision=97203 In addition, the details for the use of this change to fix the problems stated in issue #12970 may be found in the review request I made for this change. It is linked below. (closes issue #12970) Reported by: edugs15 Review: http://reviewboard.digium.com/r/116 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@184980 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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38934ec0d0 |
Additionally note that the operator option needs an 'o' extension.
(Related to issue #14731) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@183913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7e44582f57 |
Fix broken mailbox parsing when searchcontexts option is enabled.
When using the searchcontexts option in voicemail.conf, the code made the assumption that all mailbox names defined were unique across all contexts. However, the code did nothing to actually enforce this assumption, nor did it do anything to alert a user that he may have created an ambiguity in his voicemail.conf file by defining the same mailbox name in multiple contexts. With this change, we now will issue a nice long warning if searchcontexts is on and we encounter the same mailbox name in multiple contexts and ignore any duplicates after the first box. Whether searchcontexts is enabled or not, if we come across a duplicate mailbox in the same context, then we will issue a warning and ignore the duplicated mailbox. I have also added a small note to voicemail.conf.sample in the explanation for searchcontexts explaining that you cannot define the same mailbox in multiple contexts if you have enabled the option. (closes issue #14599) Reported by: lmadsen Patches: 14599.patch uploaded by mmichelson (license 60) (with slight modification) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180380 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ab5b88843c |
Clarify some documentation of queues.conf.sample
It had always been possible to explicitly specify a "blank" value for a sound file in queues.conf and have no sound played back. The problem with this is that it would result in some ugly CLI warnings from file.c. This commit introduces a check when playing a file in app_queue to see if the name of the file is zero-length and return early if that is the case. Also, the ability to specify the blank sound files in queues.conf is now mentioned more clearly in queues.conf.sample (closes issue #14227) Reported by: caspy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@180006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5996b58192 |
This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178956 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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48933e0c48 |
Add section about the #exec command in configuration files.
(closes issue #14540) Reported by: jtodd Patch by: jtodd, with additional notes by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@178445 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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13138151e1 |
Add warning to standard config, that globals may be overridden by other
dialplan configuration files. (closes issue #14388) Reported by: macli git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@173070 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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cefe4f025d |
channels/chan_dahdi.c
* Added doxygen comments to the major dahdi structures. * Fixed PRI using an incorrect string value if the extension delimiter is not present in the Dial() function. * Fixed some uninitialized string variables on FXS ports. configs/chan_dahdi.conf.sample * Updated some documentation. These changes are already in trunk -r172400 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172962 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2b340eab54 |
Rename new parkedcallparking option to parkedcallreparking
Since this option actually already existed in 1.6.0+, use the same name so as not to confuse people when they upgrade git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172639 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4e069885ce |
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags would not be set correctly for all instances when either performing a builtin attended transfer, or parking a call and getting the timeout callback. Also, there was no way on a per-call basis to specify what features someone should have on picking up a parked call (since that doesn't involve the Dial() command). There was a global option for setting whether or not all users who pickup a parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT, AUTOMON, or PARKCALL. This patch: 1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the dialplan or with setvar in channels that support it. This variable can be set to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the equivalent dial options), to set what features should be activated on this channel. The patch moves the setting of the features datastores into the bridging code instead of app_dial to help facilitate this. 2) adds global options parkedcallparking, parkedcallhangup, and parkedcallrecording to be similar to the parkedcalltransfers option for globally setting features. 3) has builtin_atxfer call builtin_parkcall if being transfered to the parking extension since tracking everything through multiple masquerades, etc. is difficult and error-prone 4) attempts to fix all cases of return calls from parking and completed builtin transfers not having the correct permissions (closes issue #14274) Reported by: aragon Patches: fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396) Tested by: aragon, otherwiseguy Review http://reviewboard.digium.com/r/138/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@172517 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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566429c300 |
Add a better explanation of the difference between the device namespace and the dialplan for newbies.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@171837 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3c19ad100a |
Remove superfluous implementation note (closes issue #14319)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170836 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8fe9b2a600 |
Add notes to the idlecheck explanation in res_odbc.conf.sample
(closes issue #14319) Reported by: klaus3000 Patches: patch_idlecheck_res_odbc.conf.sample.txt uploaded by klaus3000 (license 65) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@170719 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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464b4a8a84 |
Meetme actually has realtime but wasn't documented
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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5dca47a8bf |
s/ringdance/ringcadence/ for Bulgaria
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@168480 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a5ebe35d26 |
The documentation listed the ability to set 'maxmsg' per
context. The truth is that you can only set this in the general section or per mailbox. Thus I am updating the sample config file to be more accurate. Thanks to sasargen on IRC for bringing up this issue. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@155011 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fffb7722be |
A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying to get the one-touch parking stuff working, because it didn't occur to me that I had to also have the corresponding options in the dial command! Duh! (In all this time, I never set this up before!) So, to keep some poor fool from suffering the same fate, I made the features.conf.sample file mention the corresponding opts in dial/queue; and the docs for dial/app specifically mention the corresponding decls in the feature.conf file. I hope this doesn't spoil some vast, eternal plan... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@152538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3f6d4154b8 |
Backport fix from 1.6.0 that allows you to set parkedcalltransfers=no|caller|callee|both, but default to both which would be the equivalent of the existing behaviour.
The problem was that if someone parked a call, the callee and caller would both get assigned the builtin transfer feature, which would not only be potentially giving someone the ability to transfer themselves when they shouldn't have it, but would also dissallow reinviting the media off of the call. (closes issue #12854) Reported by: davidw Patches: parkingfix4.diff.txt uploaded by otherwiseguy Tested by: davidw, otherwiseguy git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@151763 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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1d21453b49 |
An update to the documentation/example of agents.conf.sample with the correct parameter for this feature as defined in chan_agent.c
(closes issue #13709) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@149683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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27b54f4c1c |
channels/chan_misdn.c
channels/misdn/isdn_lib.c * Miscellaneous other fixes from trunk to make merging easier later. ........ r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines * Miscellaneous formatting changes to make v1.4 and trunk more merge compatible in the mISDN area. channels/chan_misdn.c * Eliminated redundant code in cb_events() EVENT_SETUP ........ r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines improved helptext of misdn_set_opt. ........ r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line Cleaned up comment ........ r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines channels/chan_misdn.c * Made bearer2str() use allowed_bearers_array[] * Made use the causes.h defines instead of hardcoded numbers. * Made use Asterisk presentation indicator values if either of the mISDN presentation or screen options are negative. * Updated the misdn_set_opt application option descriptions. * Renamed the awkward Caller ID presentation misdn_set_opt application option value not_screened to restricted. Deprecated the not_screened option value. channels/misdn/isdn_lib.c * Made use the causes.h defines instead of hardcoded numbers. * Fixed some spelling errors and typos. * Added all defined facility code strings to fac2str(). channels/misdn/isdn_lib.h * Added doxygen comments to struct misdn_bchannel. channels/misdn/isdn_lib_intern.h * Added doxygen comments to struct misdn_stack. channels/misdn_config.c configs/misdn.conf.sample * Updated the mISDN presentation and screen parameter descriptions. doc/misdn.txt (doc/tex/misdn.tex) * Updated the misdn_set_opt application option descriptions. * Fixed some spelling errors and typos. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@145293 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a4ebc105ef |
Create rules for disallowing contacts at certain addresses, which may
improve the security of various installations. As this does not change any default behavior, it is not classified as a direct security fix for anything within Asterisk, but may help PBX admins better secure their SIP servers. (closes issue #11776) Reported by: ibc Patches: 20080829__bug11776.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, blitzrage git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@142865 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fc195a2df6 |
More fixes for realtime peers.
(closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@138258 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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960ebde9ca |
Comments in this config file were aligned only if your tab size was set to 8.
So, convert tabs to spaces so that things should be aligned regardless of what tab size you use in your editor. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@137731 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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57e3570b62 |
* The allowed_bearers setting in misdn.conf misspelled one
of its options: digital_restricted. * Fixed some other spelling errors and typos. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@136241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c981603ce4 |
fix a config sample typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@135536 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0b91843b15 |
Add a minor clarification to the documentation of mohinterpret and mohsuggest
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@135473 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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04614238c5 |
Merged revisions 132711 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@132713 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0faccba9cb |
use renamed libpri API call for controlling this feature (was improperly named before)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@132641 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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92ef406265 |
new installations should be using DAHDI instead of Zaptel, so the sample config file is now chan_dahdi.conf instead of zapata.conf
also, convert remaining references to zapata.conf in various places git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@130042 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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2ea77b3508 |
add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today
(related to issue #13042) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@130039 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fcb5675ffb |
Clear up documentation on "domain=" setting in sip.conf
Reported by: davidw (closes issue #12413) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@126844 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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84ab258c12 |
Add note about other names for EuroISDN
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@126674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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40784f499c |
Document ackcall=always.
(closes issue #12852) Reported by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@125218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3841c847a3 |
Correct description of notifyringing option.
(Closes issue #12890) Reported by gminet git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@123883 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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405dfcb54a |
Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow.
(closes issue #10417) Reported by: cstadlmann git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118646 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0ed598406b |
Add a note that pbx_config.so is needed for Local channels.
(Closes issue #12671) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@118358 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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32945354d3 |
Document exitcontext in app_voicemail sample config
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@116409 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6d2c05cbec |
Reference documentation files that actually exist.
(closes issue #12516) Reported by: linuxmaniac Patches: diff_rev114611.patch uploaded by linuxmaniac (license 472) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@114649 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |