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https://origsvn.digium.com/svn/asterisk/branches/1.8
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r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
Get real channel of a DAHDI call.
Starting with Asterisk v1.8, the DAHDI channel name format was changed for
ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
There were several reasons that the channel name had to change.
1) Call completion requires a device state for ISDN phones. The generic
device state uses the channel name.
2) Calls do not necessarily have B channels. Calls placed on hold by an
ISDN phone do not have B channels.
3) The B channel a call initially requests may not be the B channel the
call ultimately uses. Changes to the internal implementation of the
Asterisk master channel list caused deadlock problems for chan_dahdi if it
needed to change the channel name. Chan_dahdi no longer changes the
channel name.
4) DTMF attended transfers now work with ISDN phones because the channel
name is "dialable" like the chan_sip channel names.
For various reasons, some people need to know which B channel a DAHDI call
is using.
* Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and
CHANNEL(dahdi_type) so the dialplan can determine the B channel currently
in use by the channel. Use CHANNEL(no_media_path) to determine if the
channel even has a B channel.
* Added AMI event DAHDIChannel to associate a DAHDI channel with an
Asterisk channel so AMI applications can passively determine the B channel
currently in use. Calls with "no-media" as the DAHDIChannel do not have
an associated B channel. No-media calls are either on hold or
call-waiting.
(closes issue #17683)
Reported by: mrwho
Tested by: rmudgett
(closes issue #18603)
Reported by: arjankroon
Patches:
issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: stever28, rmudgett
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@309446 f38db490-d61c-443f-a65b-d21fe96a405b
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r294535 | russell | 2010-11-10 08:14:51 -0600 (Wed, 10 Nov 2010) | 5 lines
Tweak a couple of CLI commands back to their original form.
The "module" in this case is two parts, so there are two words before
the verb of the CLI command.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@294536 f38db490-d61c-443f-a65b-d21fe96a405b
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r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) | 4 lines
Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf.
Review: https://reviewboard.asterisk.org/r/922/
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r283493 | dvossel | 2010-08-24 15:34:03 -0500 (Tue, 24 Aug 2010) | 2 lines
Changes the default behavior for sip.conf's pedantic option from "no" to "yes".
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r282302 | dvossel | 2010-08-13 17:23:38 -0500 (Fri, 13 Aug 2010) | 10 lines
remove current STUN support from chan_sip.c
This patch removes the current broken/useless stun
support from chan_sip.
(closes issue #17622)
Reported by: philipp2
Review: https://reviewboard.asterisk.org/r/855/
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r281650 | russell | 2010-08-10 16:47:31 -0500 (Tue, 10 Aug 2010) | 5 lines
Change the default value for alwaysauthreject in sip.conf to "yes".
(closes issue #17756)
Reported by: oej
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281651 f38db490-d61c-443f-a65b-d21fe96a405b
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r279689 | pabelanger | 2010-07-26 19:29:34 -0400 (Mon, 26 Jul 2010) | 2 lines
Updated documentation for FAX logger level.
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r279566 | pabelanger | 2010-07-26 15:51:39 -0400 (Mon, 26 Jul 2010) | 8 lines
Add documentation for FAX logger level.
(closes issue #17715)
Reported by: vrban
Patches:
17715.patch uploaded by pabelanger (license 224)
Tested by: vrban
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