Extracting sig_analog from chan_dahdi lost call progress detection
functionality.
* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.
(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@347006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix potential deadlocks in SIP and IAX blind transfer to parking.
* Fix SIP, IAX, DAHDI analog, and MGCP channel drivers to respect the
parkext_exclusive option with transfers (Park(,,,,,exclusive_lot)
parameter). Created ast_park_call_exten() and ast_masq_park_call_exten()
to maintian API compatibility.
* Made masq_park_call() handle a failed ast_channel_masquerade() setup.
* Reduced excessive struct parkeduser.peername[] size.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@341254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Added a CLI "ss7 show channels" command that might prove useful for
future debugging.
* Made the incoming SS7 channel event check and gripe message uniform.
* Made sure that the DNID string for an incoming call is always
initialized.
(issue ASTERISK-17966)
Reported by: Kenneth Van Velthoven
Patches:
jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
NOTE: The problem was reported against v1.6.2. It is unlikely to ever
happen on v1.8 and above since chan_dahdi.c:analog_ss_thread() is unlikely
to be used. The version in sig_analog.c has largely replaced it.
(closes issue ASTERISK-18648)
Reported by: Stephan Bosch
Patches:
jira_asterisk_18648_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Stephan Bosch
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@338800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Fixes the missing DAHDI channels when using the newer chan_dahdi.conf
sections for channel configuration.
(closes issue ASTERISK-18496)
Reported by: Sean Darcy
Patches:
jira_asterisk_18496_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: Sean Darcy, rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335851 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When a SIP phone uses the dial application and receives a 484 Address
Incomplete response, if overlapped dialing is enabled for SIP, then
the 484 Address Incomplete is forwarded back to the SIP phone and the
HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete
application dialplan logic was automatically triggered; now, explicit
dialplan usage of the application is required.
Additionally, this patch adds a new AST_CONTOL_FRAME type called
AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame,
it is an indication that the dialplan expects more digits back from the
device. If the device supports overlap dialing it should attempt to
notify the device that the dialplan is waiting for more digits; otherwise,
it can handle the frame in a manner appropriate to the channel driver.
(closes issue ASTERISK-17288)
Reported by: Mikael Carlsson
Tested by: Matthew Jordan
Review: https://reviewboard.asterisk.org/r/1416/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@335064 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The following error will consistently occur when trying to dial into a
MeetMe conference when the server does not have DAHDI hardware installed:
app_meetme.c: No DAHDI channel available for conference, user introduction
disabled (is chan_dahdi loaded?)
While chan_dahdi is loaded correctly during compilation and install of
Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf
configuration file in /etc/asterisk is not created by FreePBX if hardware
does not exist, causing MeetMe to be unable to open a DAHDI pseudo
channel.
* Allow chan_dahdi to create a pseudo channel when there is no
chan_dahdi.conf file to load.
(closes issue ASTERISK-17398)
Reported by: Preston Edwards
Patches:
jira_asterisk_17398_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@334012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Change CRC4 to CRC in the output of "dahdi show status" so that it can apply in
more situations without confusing users, especially since T1 lines use CRC6
instead of CRC4.
(closes issue AST-471)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
France Telecom brings layer 2 and layer 1 down on BRI lines when the line
is idle. When layer 1 goes down Asterisk cannot make outgoing calls and
the HA8 and HB8 cards also get IRQ misses.
The inability to make outgoing calls is because the line is in red alarm
and Asterisk will not make calls over a line it considers unavailable.
The IRQ misses for the HA8 and HB8 card are because the hardware is
switching clock sources from the line which just brought layer 1 down to
internal timing.
There is a DAHDI option for the B410P card to not tell Asterisk that layer
1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp
teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards:
"modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear
up the IRQ misses when the telco brings layer 1 down.
* Add layer 2 persistence option to customize the layer 2 behavior on BRI
PTMP lines. The new option has three settings: 1) Use libpri default
layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer
brings it down. 3) Leave layer 2 down when the peer brings it down.
Layer 2 will be brought up as needed for outgoing calls.
JIRA AST-598
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Address chan_dahdi.conf dahdichan option todo item about needing line
number.
* Make ignore_failed_channels option also apply to dahdichan option.
* Don't attempt to create a default pseudo channel if the chan_dahdi.conf
channel/channels option is not allowed.
* Add a similar check for dahdichan in normal chan_dahdi.conf sections as
is done in users.conf.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* The fake event should only be processed by the channel that currently
owns the private and not the associated call waiting or 3-way channel.
JIRA AST-620
JIRA SWP-3616
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The AMI actions DAHDIHangup and DAHDITransfer have no effect on a DAHDI
channel. These two AMI actions are highly specialized to analog channels
and appear to make the channel behave like a jack port for headsets.
* Made the faked DAHDI event get processed before a normal media stream
read in dahdi_read() instead of trying to trigger an exception read by
setting the AST_FLAG_EXCEPTION flag. Apparently a change was made long
ago that changed how AST_FLAG_EXCEPTION is processed in the core.
Unfortunately, the faked DAHDI events no longer worked when that happened.
* Updated the DAHDI AMI action documentation for the following actions:
DAHDITransfer, DAHDIHangup, DAHDIDialOffhook, DAHDIDNDon, DAHDIDNDoff,
DAHDIShowChannels, and DAHDIRestart.
* Made use sscanf() instead of atoi() for better error checking of the
DAHDIChannel header string.
JIRA AST-620
JIRA SWP-3616
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The call pickup feature did not work on DAHDI devices for anything other than
feature codes beginning with * since all feature codes in chan_dahdi were
originally hard-coded to begin with *. This patch is also applied to
chan_dahdi.c to fix this bug with radio modes.
(closes issue AST-621)
Review: https://reviewboard.asterisk.org/r/1336/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The "pri show channels" command is useful for debuging to see if there are
any stuck B channels.
..........
r307964 | rmudgett | 2011-02-15 15:42:55 -0600 (Tue, 15 Feb 2011) | 9 lines
Add CLI "pri show channels" command.
List the current mapping of DAHDI B channels to Asterisk channel names and
which calls are on hold or call-waiting. Calls on hold or call-waiting
are not associated with any B channel.
JIRA LIBPRI-27
JIRA SWP-2547
..........
r308205 | rmudgett | 2011-02-17 14:21:56 -0600 (Thu, 17 Feb 2011) | 1 line
Add more verbage to CLI command 'pri show channels' usage.
..........
r312579 | rmudgett | 2011-04-04 11:17:58 -0500 (Mon, 04 Apr 2011) | 59 lines
Change also updates 'pri show channels' command with the "chan idle"
column to report if a channel is available for use.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@329012 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This matters only when autoconf fails to detect that weak linking is supported.
External optional dependencies will become optional in both cases, as they are
removed at compile time when not detected. However, runtime-optional modules
are made mandatory when weak linking is not found. This change affects only
the external optional dependencies; previously, they were incorrectly required
when weak linking support was not detected.
Patches:
20110702__issue18062__asterisk_trunk.diff.txt by tilghman (License #5003)
Tested by: iasgoscouk
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326411 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Correct ASTERISK-11323 implementation as I don't see how it ever worked as
claimed when it used the channel name and not the device name.
(issue ASTERISK-11323)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325212 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Store the defaults noted in the sample config files in the jitterbuffer config
data structure. This makes the CLI commands that output these settings show
the right thing. Also only show the settings that are relevant in the settings
CLI commands, based on which jitterbuffer is selected and whether it's enabled.
(closes issue #19083)
Reported by: rgagnon
Patches:
issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@317478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After dahdi_hangup() has supposedly hungup an ISDN channel there is still
traffic on the S0-bus because the channel was not cleaned up fully.
Shuffled the hangup code to include some missing cleanup. Also fixed some
code formatting in the area. I think the primary missing clean up code
was the call to tone_zone_play_tone() to turn off any active tones on the
channel.
(closes issue #19188)
Reported by: jg1234
Patches:
issue19188_v1.8.patch uploaded by rmudgett (license 664)
Tested by: jg1234
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@316224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Executing Dial(DAHDI/1/18475551212,300,) with the echotraining config
option enabled outputs the following debug messages unconditionally:
Dialing T1847555121 on 1
Dialing www2w on 1
* Made debug messages in my_dial_digits() normal debug messages that do
not get output unless enabled.
* Reworded some debug messages in my_dial_digits() to be clearer.
* Replace strncpy() with ast_copy_string() in my_dial_digits() which does
the same job better.
(closes issue #18847)
Reported by: vmikhelson
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
........
Merged revisions 313435 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines
reload Chan_dahdi memory leak caused by variables
chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would
stay in the dahdi_pvt structs for individual channels (causing them to just
continue adding the new ones to the list) and also there was a memory leak
causes by the conf objects. This patch resolves both of these by using
ast_variables_destroy during the loading process.
(closes issue #17450)
Reported by: nahuelgreco
Patches:
patch.diff uploaded by jrose (license 1225)
Tested by: tilghman, jrose
Review: https://reviewboard.asterisk.org/r/1170/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r313432 | jrose | 2011-04-12 13:12:29 -0500 (Tue, 12 Apr 2011) | 14 lines
reload Chan_dahdi memory leak caused by variables
chan_dahdi reloading with variables set via setvar in chan_dahdi.conf would
stay in the dahdi_pvt structs for individual channels (causing them to just
continue adding the new ones to the list) and also there was a memory leak
causes by the conf objects. This patch resolves both of these by using
ast_variables_destroy during the loading process.
(closes issue #17450)
Reported by: nahuelgreco
Patches:
patch.diff uploaded by jrose (license 1225)
Tested by: tilghman, jrose
Review: https://reviewboard.asterisk.org/r/1170/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313434 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r313189 | rmudgett | 2011-04-11 10:32:53 -0500 (Mon, 11 Apr 2011) | 32 lines
Merged revisions 313188 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r313188 | rmudgett | 2011-04-11 10:27:52 -0500 (Mon, 11 Apr 2011) | 25 lines
Stuck channel using FEATD_MF if caller hangs up at the right time.
The cause was actually a caller hanging up just at the end of the Feature
Group D DTMF tones that setup the call. The reason for this is a "guard
timer" that's implemented using ast_safe_sleep(100). If the caller
happens to hang up AFTER the final tone of the DTMF string but BEFORE the
end of that ast_safe_sleep(), then ast_safe_sleep() will return non-zero.
This causes the code to bounce to the end of ss_thread(), but it does NOT
tear down the call properly.
This should be a rare occurrence because the caller has to hang up at
EXACTLY the right time. Nonetheless, it was happening quite regularly on
the reporter's system. It's not easily reproducible, unless you purposely
increase the guard-time to 2000 or more. Once you do that, you can
reproduce it every time by watching the DTMF debug and hanging up just as
it ends.
Simply add an ast_hangup() before goto quit.
(closes issue #15671)
Reported by: jcromes
Patches:
issue15671.patch uploaded by pabelanger (license 224)
Tested by: jcromes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@313190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Regression from -r312575 B channel shifting during negotiation.
* Also combine updating the alarm flag with clearing the resetting flag.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r312574 | rmudgett | 2011-04-04 11:00:02 -0500 (Mon, 04 Apr 2011) | 45 lines
Merged revisions 312573 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r312573 | rmudgett | 2011-04-04 10:49:30 -0500 (Mon, 04 Apr 2011) | 38 lines
Issues with ISDN calls changing B channels during call negotiations.
The handling of the PROCEEDING message was not using the correct call
structure if the B channel was changed. (The same for PROGRESS.) The call
was also not hungup if the new B channel is not provisioned or is busy.
* Made all call connection messages (SETUP_ACKNOWLEDGE, PROCEEDING,
PROGRESS, ALERTING, CONNECT, CONNECT_ACKNOWLEDGE) ensure that they are
using the correct structure and B channel. If there is any problem with
the operations then the call is now hungup with an appropriate cause code.
* Made miscellaneous messages (INFORMATION, FACILITY, NOTIFY) find the
correct structure by looking for the call and not using the channel ID.
NOTIFY is an exception with versions of libpri before v1.4.11 because a
call pointer is not available for Asterisk to use.
* Made all hangup messages (DISCONNECT, RELEASE, RELEASE_COMPLETE) find
the correct structure by looking for the call and not using the channel
ID.
(closes issue #18313)
Reported by: destiny6628
Tested by: rmudgett
JIRA SWP-2620
(closes issue #18231)
Reported by: destiny6628
Tested by: rmudgett
JIRA SWP-2924
(closes issue #18488)
Reported by: jpokorny
JIRA SWP-2929
JIRA AST-437 (The issues fixed here are most likely causing this JIRA issue.)
JIRA DAHDI-406
JIRA LIBPRI-33 (Stuck resetting flag likely fixed)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@312575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines
Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround.
Not surprisingly, the workaround was exactly the same code as was provided by
the Flex maintainers, albeit in two different places, in different macros.
This should fix the FreeBSD builds, which have an older version of Flex.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@309445 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk does not send a response if we try to subscribe for call
completion after we have received a 180 Ringing. You can only subscribe
for call completion when the call has been cleared.
When we receive the 180 Ringing, for this call, its call-completion state
is 'CC_AVAILABLE'. If we then send a subscribe message to Asterisk, it
trys to change the call-completion state to 'CC_CALLER_REQUESTED'.
Because this is an invalid state change, it just ignores the message. The
only state Asterisk will accept our subscribe message is in the
'CC_CALLER_OFFERED' state.
Asterisk will go into the 'CC_CALLER_OFFERED' when the SIP client clears
the call by sending a CANCEL.
Asterisk should always send a response. Even if its a negative one.
The fix is to allow for the CCSS core to notify a CC agent that a failure
has occurred when CC is requested. The "ack" callback is replaced with a
"respond" callback. The "respond" callback has a parameter indicating
either a successful response or a specific type of failure that may need
to be communicated to the requester.
(closes issue #18336)
Reported by: GeorgeKonopacki
Tested by: mmichelson, rmudgett
JIRA SWP-2633
(closes issue #18337)
Reported by: GeorgeKonopacki
Tested by: mmichelson
JIRA SWP-2634
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@307879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines
Merged revisions 303765 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines
Sending out unnecessary PROCEEDING messages breaks overlap dialing.
Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing
through Asterisk. There is not enough information available at this point
to know if dialing is complete. The ast_exists_extension(),
ast_matchmore_extension(), and ast_canmatch_extension() calls are not
adequate to detect a dial through extension pattern of "_9!".
Workaround is to use the dialplan Proceeding() application early in
non-dial through extensions.
* Effectively revert issue #16789.
* Allow outgoing overlap dialing to hear dialtone and other early media.
A PROGRESS "inband-information is now available" message is now sent after
the SETUP_ACKNOWLEDGE message for non-digital calls. An
AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE
messages for non-digital calls.
* Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was
inconsistent with the cause codes.
* Added better protection from sending out of sequence messages by
combining several flags into a single enum value representing call
progress level.
* Added diagnostic messages for deferred overlap digits handling corner
cases.
(closes issue #17085)
Reported by: shawkris
(closes issue #18509)
Reported by: wimpy
Patches:
issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664)
Expanded upon issue18509_early_media_v1.8_v3.patch to include analog
and SS7 because of backporting requirements.
Tested by: wimpy, rmudgett
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines
Merged revisions 303284 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines
Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to users.conf
configuration. This was surprising users, and potentially setting non-sane
"defaults".
ASTNOW-125
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303467 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines
Merged revisions 303284 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines
Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to users.conf
configuration. This was surprising users, and potentially setting non-sane
"defaults".
ASTNOW-125
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@303286 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The DAHDI ISDN channel name is not dialable.
Make a channel name like DAHDI/i3/400-12 dialable when the sequence number
is stripped off of the name.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@301134 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r298194 | rmudgett | 2010-12-13 11:04:41 -0600 (Mon, 13 Dec 2010) | 26 lines
Merged revisions 298193 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r298193 | rmudgett | 2010-12-13 10:56:07 -0600 (Mon, 13 Dec 2010) | 19 lines
Outgoing PRI/BRI calls cannot do DTMF triggered transfers.
Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING
message is not received. The debug output shows that the DTMF begin event
is seen, but the DTMF end event is missing. When the DTMF begin happens,
the call is muted so we now have one way audio (until a DTMF end event is
somehow seen).
* Made set the proceeding flag when the PRI_EVENT_ANSWER event is
received.
* Made absorb the DTMF begin and DTMF end events if we are overlap dialing
and have not seen a PROCEEDING message.
* Added a debug message when absorbing a DTMF event.
JIRA SWP-2690
JIRA ABE-2697
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@298195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r296166 | rmudgett | 2010-11-24 16:42:45 -0600 (Wed, 24 Nov 2010) | 50 lines
Merged revisions 296165 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r296165 | rmudgett | 2010-11-24 16:41:07 -0600 (Wed, 24 Nov 2010) | 43 lines
Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.
The FXS connected phone has to have CW/CID support to fail, as it will
send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal
phone with no CID never fails. Also the SIP phone does not hear MOH when
the CW call is answered.
The DTMF end frame is suppressed when the phone acknowledges the CW signal
for CID. The problem is the DTMF begin frame needs to be suppressed as
well. The DTMF begin frame is causing SIP to start sending the DTMF RTP
frames. Since the DTMF end frame is suppressed, SIP will not stop sending
those DTMF RTP packets.
* Suppress the DTMF begin and end frames when the channel driver is
looking for DTMF digits.
* Fixed a couple issues caused by not cleaning up the CID spill if you
answer the CW call while it is sending the CID spill.
* Fixed not sending CW/CID spill to the phone when the call is natively
bridged. (Fixed by not using native bridge if CW/CID is possible.)
* Suppress received audio when sending CW/CID spills. The other parties
involved do not need to hear the CW/CID spills and may be confused if the
CW call is for them.
(closes issue #18129)
Reported by: alecdavis
Patches:
issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
NOTE:
* v1.4 does not have the main problem fixed by suppressing the DTMF start
frames. The other three items fixed are relevant.
* If you really must restore native bridging between analog ports, you
need to disable CW/CID either by configuring chan_dahdi.conf
callwaitingcallerid=no or dialing *70 before dialing the number to
temporarily disable CW.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@296167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Analog call waiting Caller ID spills could get stuck resulting in one
way audio until the waiting call is answered. This only happens on the
second (and later) call waiting call if the active call is not the first
call.
* The CLI/AMI "dahdi show channel" command could report the wrong channel
information.
Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer
in sync.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Restore SMDI support.
* Fixed initial value of struct analog_pvt.use_callerid. It may get
forced on depending upon other config options.
* Call analog_dnd() instead of manual inlined code.
* Removed unused struct analog_pvt.usedistinctiveringdetection.
* Removed the struct analog_pvt.unknown_alarm flag. It was really the
struct analog_pvt.inalarm flag.
* Use ast_debug() instead of ast_log(LOG_DEBUG).
* Rename several function's index variable to idx.
* Some formatting tweaks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295516 65c4cc65-6c06-0410-ace0-fbb531ad65f3
