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make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19 12:42:19 +00:00
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as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-21 20:42:37 +00:00
===
Update UPGRADE.txt and CHANGE for CDR functionality changes. Updated the UPGRADE.txt and CHANGES file stating that CDR records will not be explicity written unless cdr.conf exists and is configured. (closes issue #17373) Reported by: wdoekes Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267624 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-03 18:53:24 +00:00
=== Information for upgrading between Asterisk versions
At the request of ManxPower, include the UPGRADE.txt from 1.2 and 1.4, as well. This way, if people need to go back and review what was deprecated in previous major releases, it is readily available to them. Thanks for the suggestion! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-08 16:49:19 +00:00
===
as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-21 20:42:37 +00:00
=== These files document all the changes that MUST be taken
=== into account when upgrading between the Asterisk
=== versions listed below. These changes may require that
=== you modify your configuration files, dialplan or (in
=== some cases) source code if you have your own Asterisk
Optional HOLD/RETRIEVE signaling for PTMP TE when the bridge goes on and off hold. Added the moh_signaling option to specify what to do when the channel's bridged peer puts the ISDN channel on and off of hold. Implemented as a FSM to control libpri ISDN signaling when the bridged peer places the channel on and off of hold with the AST_CONTROL_HOLD and AST_CONTROL_UNHOLD control frames. JIRA SWP-2687 JIRA ABE-2691 Review: https://reviewboard.asterisk.org/r/1063/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-04 16:38:28 +00:00
=== modules or patches. These files also include advance
as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158449 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-21 20:42:37 +00:00
=== notice of any functionality that has been marked as
=== 'deprecated' and may be removed in a future release,
=== along with the suggested replacement functionality.
At the request of ManxPower, include the UPGRADE.txt from 1.2 and 1.4, as well. This way, if people need to go back and review what was deprecated in previous major releases, it is readily available to them. Thanks for the suggestion! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-08 16:49:19 +00:00
===
=== UPGRADE-1.2.txt -- Upgrade info for 1.0 to 1.2
=== UPGRADE-1.4.txt -- Upgrade info for 1.2 to 1.4
make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19 12:42:19 +00:00
=== UPGRADE-1.6.txt -- Upgrade info for 1.4 to 1.6
Shuffle UPGRADE.txt files for 1.10. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-23 19:17:30 +00:00
=== UPGRADE-1.8.txt -- Upgrade info for 1.6 to 1.8
Update UPGRADE.txt file for Asterisk 12 This simply pulls in the changes that were breaking from the CHANGES file and updates a few other areas accordingly. It also removes the 10 => 11 notes, which are traditionally removed from each major version and stored in the appropriate UPGRADE-X.txt file. ........ Merged revisions 398100 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30 18:38:00 +00:00
=== UPGRADE-10.txt -- Upgrade info for 1.8 to 10
=== UPGRADE-11.txt -- Upgrade info for 10 to 11
=== UPGRADE-12.txt -- Upgrade info for 11 to 12
Update UPGRADE.txt for 13 branch git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-08-08 01:33:18 +00:00
=== UPGRADE-13.txt -- Upgrade info for 12 to 13
make some corrections to the ast_agi_register_multiple(), ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19 12:42:19 +00:00
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chan_dahdi/sig_analog: Fix distinctive ring detection to suck less. The distinctive ring feature interferes with detecting Caller ID and appears to have been broken for years. What happens is if you have a ring-ring cadence as used in the UK you get too many DAHDI events for the distinctive ring pattern array and Caller ID detection is aborted. I think when Zapata/DAHDI added the ring begin event it broke distinctive ring. More events happen than before and the code does no filtering of which event times are recorded in the pattern array. * Made distinctive ring only record the ringt count when the ring ends instead of on just any DAHDI event. Distinctive ring can be ring, ring-ring, ring-ring-ring, or different ring durations for the up to three rings. * Fixed the distinctive ring detection enable (chan_dahdi.conf option usedistinctiveringdetection) to be per port instead of somewhat per port and somewhat global. This has been broken since v1.8. * Fixed using the default distinctive ring context when the detected pattern does not match any configured dringX patterns. The default context did not get set when the previous call was a matched distinctive ring pattern and the current call is not matched. This has been broken since v1.8. * Made distinctive ring have no effect on Caller ID detection when it is disabled. Caller ID detection just monitors for 10 seconds before giving up. * Fixed leak of struct callerid_state memory when a polarity reversal during Caller ID detection causes the incoming call to be aborted. DAHDI-1143 AST-1545 ASTERISK-24825 #close Reported by: Richard Mudgett ASTERISK-17588 Reported by: Daniel Flounders Review: https://reviewboard.asterisk.org/r/4444/ ........ Merged revisions 432530 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432534 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432551 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-06 20:24:58 +00:00
Channel Drivers:
chan_dahdi:
- For users using the FXO port (FXS signaling) distinctive ring detection
feature, you will need to adjust the dringX count values. The count
values now only record ring end events instead of any DAHDI event. A
ring-ring-ring pattern would exceed the pattern limits and stop
Caller-ID detection.
res_http_websocket: Close websocket correctly and use careful fwrite When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. Note that this version of the patch, unlike r417310 in Asterisk 11, exposes configuration options beyond just chan_sip's sip.conf. Configuration options to configure the write timeout have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ ........ Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-26 12:21:14 +00:00
Logger: Convert 'struct ast_callid' to unsigned int. Switch logger callid's from AO2 objects to simple integers. This helps in two ways. Copying integers is faster than referencing AO2 objects, so this will result in a small reduction in logger overhead. This also erases the possibility of an infinate loop caused by an invalid callid in threadstorage. ASTERISK-24833 #comment Committed callid conversion to trunk. Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4466/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-03-13 01:12:35 +00:00
Core:
Logging:
- The first callid created is now 1 instead of 0. The value 0
is now reserved to represent a lack of callid.
Shuffle UPGRADE.txt files for 1.10. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@279118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-23 19:17:30 +00:00
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