https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55006 | russell | 2007-02-16 16:49:42 -0600 (Fri, 16 Feb 2007) | 17 lines
Merged revisions 55005 via svnmerge from
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r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines
Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4,
and trunk. I decided that once a conference is created from meetme.conf,
it is acceptable behavior that the pin can not be changed until the
conference goes away. I also added a note in meetme.conf to describe this
behavior.
We still have another issue in 1.4 and trunk where some conferences with no
users don't go away. That is the real bug that needs to be addressed here.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55002 | file | 2007-02-16 17:18:46 -0500 (Fri, 16 Feb 2007) | 10 lines
Merged revisions 54999 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54999 | file | 2007-02-16 17:13:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Do not send indications through ast_indicate in chan_agent but instead go directly to the technology. This way when indications are emulated they happen on the Agent channel and do not screw up formats on the channels. (issue #8439 reported by punkgode)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54969 | russell | 2007-02-16 15:12:18 -0600 (Fri, 16 Feb 2007) | 13 lines
Merged revisions 54955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | 5 lines
For conferences that are configured in meetme.conf, check the configuration
file every time someone joins the conference instead of only when the
conference is first created. This is to ensure that changes to the pin
numbers in the config file are always honored. (issue #9073)
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r54898 | russell | 2007-02-16 12:03:41 -0600 (Fri, 16 Feb 2007) | 4 lines
Fix setting "autofallthrough" to yes by default. It was set to enabled in
pbx.c. However, if the option was not present in extensions.conf, then
pbx_config.c would set it back to disabled.
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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54838 65c4cc65-6c06-0410-ace0-fbb531ad65f3
pretty cool things.
First, you can get the device state of anything in the dialplan:
NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)})
NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)})
Most importantly, this allows you to create custom device states so you can
control phone lamps directly from the dialplan.
Set(DEVSTATE(Custom:mycustomlamp)=BUSY)
...
exten => mycustomlamp,hint,Custom:mycustomlamp
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@54261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54204 | russell | 2007-02-13 13:42:00 -0600 (Tue, 13 Feb 2007) | 5 lines
If we fail to create the SIP socket, then return -1 from reload_config() so
that load_module() will return AST_MODULE_LOAD_DECLINE. Otherwise, the console
will just get spammed with error messages every time chan_sip tries to send a
message.
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r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines
- Add the ability to register a callback to monitor state changes in an
asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API
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