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
Before, having multiple subscriptions to mailboxes on a sip peer set via the mailbox
setting in sip.conf would only result in updates being sent on whichever mailbox
triggered the mwi event. Now all of them get counted regardless. Also fixes a bug
involving parsing of the mailbox option in sip.conf so that trailing and leading
spaces before/after commas are trimmed.
(closes issue ASTERISK-18067)
Reported by: aragon
(closes issue ASTERISK-15479)
Reported by: Ben Winslow
Patches: chan_sip.c-mwi_multi_mailbox_fix-1.6.2.13.diff (License #5288) patch uploaded by Ben Winslow
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332118 65c4cc65-6c06-0410-ace0-fbb531ad65f3
setting of HASH(SIP_CAUSE,<chan name>) on the channel.
Having chan_sip set HASH(SIP_CAUSE,<chan name>) on the channel carries a
significant performance penalty because of the usage of the MASTER_CHANNEL()
dialplan function.
AST-580
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@332021 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
Any SIP notify sent via AMI or CLI leaks a SIP PVT with ref count +2. Removing
the additional ref just before the invite and adding an unref following it
corrects the issue as seen via REF_DEBUG. The unref existed in a distant
revision and it appears as though the wrong ref operation was removed.
(closes issue ASTERISK-18091)
Review: https://reviewboard.asterisk.org/r/1332/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add some reentrancy protection in pbx.c when creating the contexts_table
hash table.
* Fix inverted test in chan_sip.c conditional code.
* Fix uninitialized variable and use of the wrong variable in chan_iax2.c.
* Fix test of return value in app_parkandannounce.c. Explicitly testing
for -1 is bad if the function does not actually return that value when it
fails.
* Fixup some comments and add some curly braces in features.c.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@331248 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
Fixes crash in chan_iax2 resulting from an edge case in the
way control frames are queued during calltoken negotiation is complete.
(closes issue ASTERISK-17610)
Reported by: mgrobecker
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
..........
r330033 | rmudgett | 2011-07-28 11:26:38 -0500 (Thu, 28 Jul 2011) | 15 lines
Datacalls with B410P fail.
Incoming and outgoing call legs of a data call are using different
formats: a-law, u-law. When the call is bridged, the media stream is run
through translation to convert the media formats. The translation is bad
for data calls.
* Make incoming call that does not explicitly specify u-law or a-law use
the DAHDI channel's default law. The outgoing call always uses the
default law from the DAHDI channel.
(closes issue ABE-2800)
Patches:
jira_abe_2800_companding.patch (license #5621) patch uploaded by rmudgett
..........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@330050 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
The functionality of inband DTMF in chan_sip relied upon
ast_rtp_instance_dtmf_mode_get/set not working properly to avoid calling
ast_rtp_instance_dtmf_begin/end on RTP streams with inband DTMF. According to
documentation, ast_rtp_instance_dtmf_begin/end is meant only for RFC2833 DTMF,
never inband. This fixes the regression introduced in revision 328823.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When deciding whether Asterisk was allowed to bridge the call away from the
core, chan_sip did not take into account the usage of features on dialed
channels that require monitoring of DTMF on channels utilizing inband DTMF.
This would cause Asterisk to allow the call to be locally or remotely bridged,
preventing access to the data required to detect activations of such features.
(closes 17237)
Review: https://reviewboard.asterisk.org/r/1302/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Ideally, sip_setoption shouldn't be called if there is a lack of a sip private structure. But this will fix a crash.
(closes issue ASTERISK-17909)
Reported by: Mark Murawski
Tested by: Mark Murawski
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@328608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk sends a 403 Forbidden response if authentication fails for an
INVITE as required. However, it ignores the ACK and keeps retransmitting
the response.
* Made not delete the to-tag in the dialog so the expected ACK can be
matched with the dialog and stop the retransmissions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@327211 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
If you use the auth= parameter and do a "sip reload" while there is an
ongoing call. The peer->auth data points to free'd memory.
The patch does several things:
1) Puts the authentication list into an ao2 object for reference counting
to fix the reported crash during a SIP reload.
2) Converts the authentication list from open coding to AST list macros.
3) Adds display of the global authentication list in "sip show settings".
(closes issue ASTERISK-17939)
Reported by: wdoekes
Patches:
jira_asterisk_17939_v1.8.patch (license #5621) patch uploaded by rmudgett
Review: https://reviewboard.asterisk.org/r/1303/
JIRA SWP-3526
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Redoes -r308945 for issue ASTERISK-17431 deadlock fix for
sip_set_udptl_peer() and sip_set_rtp_peer().
* Lock the channels in the defined order and avoid the need for a deadlock
avoidance loop.
* Lock the channel before getting the pointer to the private structure to
be sure that the pointer will not change due to a masquerade or channel
hangup.
* To preserve sanity, check that chan and p->owner are the same. (Pointer
rearangements should not happen without the protection of locks because
bad things tend to happen otherwise.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@326144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Add load failure exit if primary SIP container(s) could not get created
in chan_sip.c:load_module().
* Removed a redundant static prototype.
* Some typos.
* Some whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@325935 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
A remote peer subscribed to MWI with the unsolicited option and a local
phone subscribed to the remote mailbox. The notify message-summary events
are sent correctly except for the first one when subscribing, which will
always be 0. This means the phone MWI indicator will be wrong until the
mailbox read/unread count changes and the event is fired.
Looks like this is a regression from ASTERISK-16149.
* Fix the logic to check the cache and if allowed then fallback to
manually counting mailbox messages.
(closes issue ASTERISK-17997)
Reported by: rsw686
Patches:
jira_asterisk_17997_v1.8.patch (license #5621) uploaded by rmudgett
Tested by: rsw686
JIRA SWP-3551
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@324914 65c4cc65-6c06-0410-ace0-fbb531ad65f3
