Age | Commit message (Collapse) | Author |
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remote party on incoming calls from the pstn, this shouldnt be enabled, only if the provider allows it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10225 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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#6315, plus documentation)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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asterisk standard, changed it so it can take frames
of sizes other than 20ms, allowed for the app to reload
properly, and finalyl changed sample to general section
as to follow standards.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10086 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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fixes. Thanks to all those involved.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9918 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Moving the timezone section to below the documentation section
Example written and contributed by Henrik Olsen, Astricon Training student.
Thanks!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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0, this fixes the tv_fix warnings, because we use ast_read to transmit frames to asterisk in jitterbuffer mode, instead of queueing the audio data with ast_queue_frame.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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* fixed "RETRIEVE does not work" bug
* fixed some NT Mode bugs
* removed some // comments
* added configureable jitterbuffer
* removed own tone-generator, and use asterisks instead, to support
asterisks indications
* added more support for hw-bridging, we bridge now every possible call
* fixed some hdlc mode issues, with a patch for chan_zap we can make
data calls between chan_zap and chan_misdn now
* completely reworked the config engine, works like a charm now
* fixed SetCallerPres - bug
* added Progress and Proceeding passing
* optimized Ringing Indication handling
* added full ast_send_text support (you can setup nice menus with the dialplan
now)
* added support to read /etc/misdn-init.conf to clarify the NT+PTP Problem
* we compile now channels/misdn if mISDNuser is installed systemwide
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9114 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Moving files from / to /doc or /configs
- Renaming some documentation files
Thank you for the initiative, manxpower!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Add some extra headers and reference to other doc/ files for realtime
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8926 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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fourcheeze in the IRC channel
for pointing this out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(inspired by bug #6183)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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cleaning patch 6267
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8495 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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- Add callingpres to sip.conf
- Add reference to README.callingpres from zapata.conf
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8336 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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compatible
with other channels.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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monitor app from manager and in the call through
features.conf bug 5395 for the patch
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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mods)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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autopause. thanks twisted from 5577
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8060 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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other part.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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as multiple agents at same time. bug 6046 thanks
ppyy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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fixes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8001 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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/branches/1.2
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- Add documentation on call-limit, explaining that there's two counters
for a type="friend".
- Document the removval of "incominglimit" in UPGRADE.txt
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7775 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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of including commas (for LDAP configuration)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7642 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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for Feature Group D Tamdem trunks inbound (outbound was already there)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7599 | russell | 2005-12-22 15:36:47 -0500 (Thu, 22 Dec 2005) | 3 lines
revert changes to videosupport to allow per-peer setting, since it isn't quite
complete and there is not an obvious fix at this point
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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with "Touch monitor"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7405 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7285 | tilghman | 2005-12-02 15:12:05 -0600 (Fri, 02 Dec 2005) | 2 lines
Turn on executable bits for startup scripts, and fix bash var interpolation for Mandrake
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r7299 | oej | 2005-12-02 19:24:40 -0600 (Fri, 02 Dec 2005) | 2 lines
Documenting the default registerattempts setting as 0, continue hammering the server for ever and ever ;-)
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r7310 | tilghman | 2005-12-03 13:55:05 -0600 (Sat, 03 Dec 2005) | 3 lines
Bug 5925: check for "Unknown", as that's what app_voicemail puts into the field for Unknown callerid
Also, remove useless res checks (initialized to 0; never set)
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r7329 | kpfleming | 2005-12-04 12:03:07 -0600 (Sun, 04 Dec 2005) | 2 lines
use a more efficient way to get the revision number, that will also report if the working copy contains uncommitted modifications
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r7265 | oej | 2005-12-01 17:18:14 -0600 (Thu, 01 Dec 2005) | 2 lines
Changing bug report address to the Asterisk issue tracker
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r7266 | kpfleming | 2005-12-01 17:18:29 -0600 (Thu, 01 Dec 2005) | 3 lines
Makefile 'update' target now supports updating from Subversion repositories (issue #5875)
remove support for 'patches' subdirectory, it's no longer useful
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r7268 | kpfleming | 2005-12-01 17:34:58 -0600 (Thu, 01 Dec 2005) | 2 lines
ensure channel's scheduling context is freed (issue #5788)
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r7269 | kpfleming | 2005-12-01 17:49:44 -0600 (Thu, 01 Dec 2005) | 2 lines
don't block waiting for the Festival server forever when it goes away (issue #5882)
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r7270 | kpfleming | 2005-12-01 18:26:12 -0600 (Thu, 01 Dec 2005) | 2 lines
allow variables to exist on both 'halves' of the Local channel (issue #5810)
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r7271 | kpfleming | 2005-12-01 18:28:48 -0600 (Thu, 01 Dec 2005) | 2 lines
protect agent_bridgedchannel() from segfaulting when there is no bridged channel (issue #5879)
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r7272 | kpfleming | 2005-12-01 18:39:00 -0600 (Thu, 01 Dec 2005) | 3 lines
properly handle password changes when mailbox is last line of config file and not followed by a newline (issue #5870)
reformat password changing code to conform to coding guidelines (issue #5870)
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r7273 | kpfleming | 2005-12-01 18:42:40 -0600 (Thu, 01 Dec 2005) | 2 lines
allow previous context-searching behavior to be used if desired (issue #5899)
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r7274 | kpfleming | 2005-12-01 18:51:15 -0600 (Thu, 01 Dec 2005) | 2 lines
inherit channel variables into channels created by Page() application (issue #5888)
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r7275 | oej | 2005-12-01 18:52:13 -0600 (Thu, 01 Dec 2005) | 2 lines
Bug #5907. Improve SIP INFO DTMF debugging output. (1.2 & Trunk)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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