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2009-10-05Merged revisions 222110 via svnmerge from kpfleming1-0/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r222110 | kpfleming | 2009-10-05 14:45:00 -0500 (Mon, 05 Oct 2009) | 25 lines Allow non-compliant T.38 endpoints to be supportable via configuration option. Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept as the T38FaxMaxDatagram value in their SDP, when in fact this value is supposed to be the maximum UDPTL payload size (datagram size) they can accept. If the value they supply is small enough (a commonly supplied value is '72'), T.38 UDPTL transmissions will likely fail completely because the UDPTL packets will not have enough room for a primary IFP frame and the redundancy used for error correction. If this occurs, the Asterisk UDPTL stack will emit log messages warning that data loss may occur, and that the value may need to be overridden. This patch extends the 't38pt_udptl' configuration option in sip.conf to allow the administrator to override the value supplied by the remote endpoint and supply a value that allows T.38 FAX transmissions to be successful with that endpoint. In addition, in any SIP call where the override takes effect, a debug message will be printed to that effect. This patch also removes the T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not actually had any effect for a number of releases. In addition, this patch cleans up the T.38 documentation in sip.conf.sample (which incorrectly documented that T.38 support was passthrough only). (issue #15586) Reported by: globalnetinc ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@222112 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-01Merged revisions 221592 via svnmerge from kpfleming1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r221592 | kpfleming | 2009-10-01 11:16:09 -0500 (Thu, 01 Oct 2009) | 12 lines Remove ability to control T.38 FAX error correction from udptl.conf. chan_sip has had the ability to control T.38 FAX error correction mode on a per-peer (or global) basis for a couple of releases now, which is where it should have been all along. This patch removes the ability to configure it in udptl.conf, but issues a warning if the user tries to do, telling them to look at sip.conf.sample for how to configure it now. For any SIP peers that are T.38 enabled in sip.conf, there is already a default for FEC error correction even if the user does not specify any mode, so this change will not turn off error correction by default, it will have the same default value that has been in the udptl.conf sample file. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@221602 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-16Merged revisions 218799 via svnmerge from russell1-0/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r218799 | russell | 2009-09-16 08:34:41 -0500 (Wed, 16 Sep 2009) | 16 lines Merged revisions 218798 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r218798 | russell | 2009-09-16 08:33:43 -0500 (Wed, 16 Sep 2009) | 9 lines Remove the IAXy firmware from Asterisk. The firmware can now be found on downloads.digium.com, where the rest of our binary downloads live. This was the last part of our Asterisk tarballs that was considered non-free by Debian. :-) (closes issue #15838) Reported by: paravoid ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@218801 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03tweakrussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216098 f38db490-d61c-443f-a65b-d21fe96a405b
2009-09-03Merged revisions 216092 via svnmerge from russell1-0/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r216092 | russell | 2009-09-03 14:38:35 -0500 (Thu, 03 Sep 2009) | 16 lines Merged revisions 216085 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r216085 | russell | 2009-09-03 14:36:46 -0500 (Thu, 03 Sep 2009) | 9 lines Merged revisions 216080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r216080 | russell | 2009-09-03 14:35:23 -0500 (Thu, 03 Sep 2009) | 2 lines Add a note about IAX2 to UPGRADE.txt. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@216095 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-23Use correct formatting for T.38 change note in UPGRADE.txtkpfleming1-9/+9
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@208503 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-23Merged revisions 208464 via svnmerge from kpfleming1-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ........ r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines Rework of T.38 negotiation and UDPTL API to address interoperability problems Over the past couple of months, a number of issues with Asterisk negotiating (and successfully completing) T.38 sessions with various endpoints have been found. This patch attempts to address many of them, primarily focused around ensuring that the endpoints' MaxDatagram size is honored, and in addition by ensuring that T.38 session parameter negotiation is performed correctly according to the ITU T.38 Recommendation. The major changes here are: 1) T.38 applications in Asterisk (app_fax) only generate/receive IFP packets, they do not ever work with UDPTL packets. As a result of this, they cannot be allowed to generate packets that would overflow the other endpoints' MaxDatagram size after the UDPTL stack adds any error correction information. With this patch, the application is told the maximum *IFP* size it can generate, based on a calculation using the far end MaxDatagram size and the active error correction mode on the T.38 session. The same is true for sending *our* MaxDatagram size to the remote endpoint; it is computed from the value that the application says it can accept (for a single IFP packet) combined with the active error correction mode. 2) All treatment of T.38 session parameters as 'capabilities' in chan_sip has been removed; these parameters are not at all like audio/video stream capabilities. There are strict rules to follow for computing an answer to a T.38 offer, and chan_sip now follows those rules, using the desired parameters from the application (or channel) that wants to accept the T.38 negotiation. 3) chan_sip now stores and forwards ast_control_t38_parameters structures for tracking 'our' and 'their' T.38 session parameters; this greatly simplifies negotiation, especially for pass-through calls. 4) Since T.38 negotiation without specifying parameters or receiving the final negotiated parameters is not very worthwhile, the AST_CONTROL_T38 control frame has been removed. A note has been added to UPGRADE.txt about this removal, since any out-of-tree applications that use it will no longer function properly until they are upgraded to use AST_CONTROL_T38_PARAMETERS. Review: https://reviewboard.asterisk.org/r/310/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@208484 f38db490-d61c-443f-a65b-d21fe96a405b
2009-07-17Recorded merge of revisions 207224 via svnmerge from tilghman1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r207224 | tilghman | 2009-07-17 17:04:43 -0500 (Fri, 17 Jul 2009) | 2 lines Document the "flag" field in the voicemessages table. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@207227 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-30Merged revisions 204563 via svnmerge from tilghman1-2/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r204563 | tilghman | 2009-06-30 15:41:04 -0500 (Tue, 30 Jun 2009) | 13 lines Merged revisions 204556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204556 | tilghman | 2009-06-30 15:23:51 -0500 (Tue, 30 Jun 2009) | 6 lines More incorrect language codes, plus ensuring that regionalizations use the specified language, and not English for grammar. (closes issue #15022) Reported by: greenfieldtech Patches: 20090519__issue15022.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@204612 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-30Recorded merge of revisions 204470 via svnmerge from tilghman1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines Recorded merge of revisions 204469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines "tw" is the language specification for Twi (from Ghana) not Taiwanese. (closes issue #15346) Reported by: volivier Patches: 20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14) 20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14) Tested by: volivier ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@204472 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Merged revisions 179154 via svnmerge from russell1-0/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r179154 | russell | 2009-02-27 15:23:12 -0600 (Fri, 27 Feb 2009) | 2 lines Add a note about the ordering of entries in sip.conf in 1.6.1. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@179160 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-25Merged revisions 158924 via svnmerge from mnicholson1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r158924 | mnicholson | 2008-11-24 18:05:41 -0600 (Mon, 24 Nov 2008) | 6 lines Make the Join event from app_queue use CallerIDNum insead of CallerID for indicating the callerid number just like the rest of asterisk. (closes issue #13883) Reported by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@158927 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-21Merged revisions 158449 via svnmerge from kpfleming1-0/+11
https://origsvn.digium.com/svn/asterisk/trunk ........ r158449 | kpfleming | 2008-11-21 14:42:37 -0600 (Fri, 21 Nov 2008) | 3 lines as suggested by jtodd, document the purposes of the CHANGES and UPGRADE files ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@158450 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-19Merged revisions 157706 via svnmerge from kpfleming1-272/+33
https://origsvn.digium.com/svn/asterisk/trunk ........ r157706 | kpfleming | 2008-11-19 06:42:19 -0600 (Wed, 19 Nov 2008) | 5 lines 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@157719 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-13Merged revisions 137530 via svnmerge from kpfleming1-1/+25
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug 2008) | 1 line add document describing what users will need to be aware of when upgrading to this version and using DAHDI ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137627 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-11Log the userfield CDR variable like the other CDR backends, assuming theseanbright1-1/+2
column is actually there. If it's not, we still log everything else as before. (closes issue #13281) Reported by: falves11 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@137203 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-05Add '+=' append operator to configuration files.tilghman1-0/+7
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@135717 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-28Change SendImage() to output a more consistent status variable.tilghman1-0/+7
(closes issue #13134) Reported by: eliel Patches: app_image.c.patch uploaded by eliel (license 64) UPGRADE.patch uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@134088 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-17Change several 'core' commands to be 'dialplan' commands (with appropriatetilghman1-0/+5
deprecation, of course) (closes issue #13016) Reported by: caio1982 Patches: dialplan_globals6.diff uploaded by caio1982 (license 22) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@131606 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-28Merge in changes from my cdr-tds-conversion branch. This changes the internalseanbright1-0/+3
implementation from using the volatile libtds, to using the db-lib front end. The unintended side effect of this is that we support (at least) versions 0.62 through 0.82 of the FreeTDS distribution without any #ifdef ugliness. (closes issue #12844) Reported by: jcollie git-svn-id: http://svn.digium.com/svn/asterisk/trunk@126226 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-24Update CHANGES and UPGRADE.txt per kpfleming's mail to #asterisk-dev.seanbright1-4/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124835 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-15Add some more IAX2-specific information about the channel to the CHANNEL()tilghman1-0/+3
function and begin the transition from SIPCHANINFO() to just using CHANNEL(). (closes issue #12856) Reported by: mostyn Patches: iax_and_sip_channel_info.patch uploaded by mostyn (license 398) (with some additional cleanup by me) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122802 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-11Expand CDR uniqueid field to 150 chars, to account for maximum systemname.tilghman1-0/+5
(Closes issue #12831) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@121855 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-05Add info on the [compat] section of asterisk.conf.tilghman1-1/+15
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120567 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-25Change space-zero to now evaluate to false, as is expected by a great many.tilghman1-0/+6
(Inspired by a post on the -users list) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@118223 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-15Modify externnotify to take the number of urgent voicemails as a final ↵mmichelson1-1/+1
argument instead of the string "Urgent" (closes issue #12660) Reported by: jaroth Patches: externnotify.patch uploaded by jaroth (license 50) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116592 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-09Adding support for "urgent" voicemail messages. Messages which aremmichelson1-0/+2
marked "urgent" are considered to be higher priority than other messages and so they will be played before any other messages in a user's mailbox. There are two ways to leave an urgent message. 1. send the 'U' option to VoiceMail(). 2. Set review=yes in voicemail.conf. This will give instructions for a caller to mark a message as urgent after the message has been recorded. I have tested that this works correctly with file and ODBC storage, and James Rothenberger (who wrote initial support for this feature) has tested its use with IMAP storage. (closes issue #11817) Reported by: jaroth Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent Tested by: putnopvut, jaroth git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115588 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-05Note change for ExecIf syntax (caught by jmls on IRC)tilghman1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115302 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-05clarify wordingkpfleming1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115288 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-01Clarify the deprecation notice about Macro() to note that it will not be removedrussell1-1/+6
for the sake of backwards compatibility, since it is a non-trivial task to convert existing large dialplans that depend on Macro() to use GoSub(), instead. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114931 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-29Make app_directory dependent on app_voicemail. This is because the functionmmichelson1-0/+2
which says the person's name is handled inside app_voicemail now. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114841 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-26Merged revisions 111126 via svnmerge from kpfleming1-12/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines update UPGRADE notes to document usage of the script ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@111127 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-26Merged revisions 110962 via svnmerge from kpfleming1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines add note that the user will need to enable codec_ilbc to get it to build ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110963 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-26Merged revisions 110880 via svnmerge from kpfleming1-0/+18
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@110881 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-05Create a centralized configuration option for silencethresholdtilghman1-0/+3
(closes issue #11236) Reported by: philipps Patches: 20080218__bug11236.diff.txt uploaded by Corydon76 (license 14) Tested by: philipps git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106072 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-25Deprecate the "stripmsd" option in favor of dialplan substring variable syntax.russell1-0/+3
(closes issue #12060) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104110 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-22Move Originate to a separate privilege and require the additional System ↵tilghman1-0/+3
privilege to call out to a subshell. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@104039 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-19Add CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan ↵file1-0/+3
application. This will either be set to NOCHANNEL if the given channel was not found or SUCCESS if it worked. (closes issue #11553) Reported by: johan Patches: UPGRADE.txt.channelredirect.patch uploaded by johan (license 334) CHANGES.channelredirect.patch uploaded by johan (license 334) app_channelredirect-20080219.patch uploaded by johan (license 334) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103819 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-14Change the queue holdtime announcement to happen at any interval (not just ↵mmichelson1-0/+3
greater than two minutes). Remove the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using '1' as a queue-round-seconds value is no longer valid. (closes issue #9736) Reported by: caio1982 Patches: queue_announce5.diff uploaded by caio1982 (license 22) Tested by: caio1982, putnopvut git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103687 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-131. Deprecate SetMusicOnHold and WaitMusicOnHold.mmichelson1-0/+5
2. Add a duration parameter to MusicOnHold (closes issue #11904) Reported by: dimas Patches: v2-moh.patch uploaded by dimas (license 88) Tested by: dimas git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103658 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-08At the request of ManxPower, include the UPGRADE.txt from 1.2 and 1.4, as well.russell1-1/+7
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: http://svn.digium.com/svn/asterisk/trunk@103044 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-05Add a note about changing modules.conf since another console channel driver isrussell1-0/+7
now present that can not be used at the same time as chan_alsa or chan_oss. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96715 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-16HUGE improvements to QoS/CoS handling by IgorGoej1-0/+3
- Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93163 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14Remove use of privacy.conf by the Privacy app.tilghman1-0/+2
Reported by: eliel Patch by: eliel (Closes issue #11344) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@93066 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-06Adding documentation for the massive manager changes to manageroej1-3/+2
version 1.1 - hopefully a more consistent manager interface. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91438 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Change cdr_manager to use a "CDR" level, rather than the (overcrowded) ↵tilghman1-0/+4
"call" level. (Closes issue #11015) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91173 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Rename "username" to "defaultuser" to match with "defaultip".oej1-0/+5
"Username" still works, but is deprecated. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91152 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26- Mark "concise" as deprecatedoej1-2/+9
- Restructure other changes to UPGRADE.txt and CHANGES We're still looking for scripts that replace asterisk -rx "show shannels concise" by using the manager interface, but still produces the same output. Anyone? git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89606 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21Change Read to set READSTATUS as an indication of the resulttilghman1-0/+3
Also, some cleanup to CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman (Closes issue #11004) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89489 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-15Fix typo in UPGRADE.txt. 'increase' should have been used, not 'increasing'.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89291 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-23 18:34:26 +0000