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The CLI "dahdi show channel" command was not correctly reporting the
InAlarm status.
The inalarm flag is now consistently passed between chan_dahdi and
submodules.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261007 f38db490-d61c-443f-a65b-d21fe96a405b
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Created
SIG_PRI_MAX_CHANNELS, SIG_PRI_NUM_DCHANS
SIG_SS7_MAX_CHANNELS, SIG_SS7_NUM_DCHANS
SIG_MFCR2_MAX_CHANNELS
Also fixed the declaration of pollers[] in mfcr2_monitor(). It was
dimensioned to the number of bytes in struct dahdi_mfcr2.pvts[] and not to
the same dimension of the struct dahdi_mfcr2.pvts[].
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260435 f38db490-d61c-443f-a65b-d21fe96a405b
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From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
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Also avoid potential crash because cause could be NULL.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@251585 f38db490-d61c-443f-a65b-d21fe96a405b
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Rework HAVE_PRI_SERVICE_MESSAGES to not use the active values directly
from the database. Database access is likely expensive. Database access
now only happens on initialization, destruction, and when the B channel is
taken in or out of service.
This change is not related to call waiting but it would cause the search
for a call waiting interface to be very expensive and slow down D channel
message servicing.
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Changed areas in sig_pri to set the digital flag using a callback that will
also set the corresponding flag in chan_dahdi. Modified dahdi_request slightly
so that if a bearer is marked as digital, that information is available when
creating the new channel.
(closes issue #16151)
Reported by: alecdavis
Patch based on bug_16151.diff.txt uploaded by alecdavis (license 585)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@231058 f38db490-d61c-443f-a65b-d21fe96a405b
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solution.)
Since ISDN works like SIP and not analog ports in regard to devices, the
device state based on the ISDN channel number could not work. This has
not been an issue until the advent of PTMP NT mode. Previously, ISDN
lines were used as trunks and did not have to keep track of specific
devices.
As an interim solution until device states are properly implemented, the
channel name is being changed to the following format to use the generic
device state support:
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>
Dialplan hints would thus be:
exten => xxx,hint,DAHDI/i2/5551212
This will work with the following restrictions:
* The number of devices/phones cannot exceed the number of B channels.
(i.e., BRI has 2)
* Each device/phone can only have one number. No shared MSN's.
* The phones/devices probably should not use subaddressing.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@226882 f38db490-d61c-443f-a65b-d21fe96a405b
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* Added handling of received HOLD/RETRIEVE messages and the optional ability
to transfer a held call on disconnect similar to an analog phone.
* Added CallRerouting/CallDeflection support for Q.SIG, ETSI PTP, ETSI PTMP.
Will reroute/deflect an outgoing call when receive the message.
Can use the DAHDISendCallreroutingFacility to send the message for the
supported switches.
* Added ability to send/receive keypad digits in the SETUP message.
Send keypad digits in SETUP message: Dial(DAHDI/g1[/K<keypad_digits>][/extension])
Access any received keypad digits in SETUP message by: ${CHANNEL(keypad_digits)}
* Added support for BRI PTMP NT mode.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@225692 f38db490-d61c-443f-a65b-d21fe96a405b
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control frame on a channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@224491 f38db490-d61c-443f-a65b-d21fe96a405b
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The PRI channels can no longer change the channel name if a different B
channel is selected during call negotiation. To prevent using the channel
name to infer what B channel a call is using and to avoid name collisions,
the channel name format is changed.
The new channel naming for PRI channels is:
DAHDI/ISDN-<span>-<sequence-number>
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221701 f38db490-d61c-443f-a65b-d21fe96a405b
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* Removed unneeded and uninitialized service_lock.
* Fixed potential locking imbalance in pri_dchannel():PRI_EVENT_RESTART.
* Fixed verbose message typo in pri_dchannel():PRI_EVENT_RESTART.
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ISDN PTMP CPE spans.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@215757 f38db490-d61c-443f-a65b-d21fe96a405b
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Add Connected Line Presentation (COLP) support to chan_dahdi/libpri as an
addition to issue 8824. This is the chan_dahdi/sig_pri portion. COLP
support is now available for any switch for which libpri supports COLP
(currently ETSI PTP, ETSI PTMP, and Q.SIG) with this patch.
(closes issue #14068)
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/340/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@213007 f38db490-d61c-443f-a65b-d21fe96a405b
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Putting some DSP code back into sig_pri that was removed by the
chan_dahdi/sig_pri reorganization.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211392 f38db490-d61c-443f-a65b-d21fe96a405b
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* Q.SIG channel mapping option.
* discardremoteholdretrieval option.
* libPRI debug defines.
* pri_set_overlapdial() now set correctly.
* pthread creation of pri_ss_thread now matches.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@210696 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15614)
Reported by: fabled
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209619 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #15595)
Reported by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@209554 f38db490-d61c-443f-a65b-d21fe96a405b
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The wrong logic was being used in chan_dahdi to convert a sig_pri_chan
to the proper libpri channel number. The most significant bit must only
be set only when trunk groups are being used.
(closes issue #15452)
Reported by: alecdavis
Patches:
bug15452.patch uploaded by jpeeler (license 325)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208267 f38db490-d61c-443f-a65b-d21fe96a405b
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This readds the proper setting of the flag and is really a continuation of
r205731. The flag was being set properly in sig_analog, but use of the
newly added set_dialing callback allowed for some simplification in
chan_dahdi.
(closes issue #15486)
Reported by: rmudgett
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@206767 f38db490-d61c-443f-a65b-d21fe96a405b
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Also go back and wrap all of the places that use the specific reverse charge
APIs with preprocessor conditionals.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204919 f38db490-d61c-443f-a65b-d21fe96a405b
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This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI. This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:
...,Dial(DAHDI/g1/C4445556666)
And to read it off an inbound channel:
exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
Thanks again to rmudgett for the thorough review.
(closes issue #13760)
Reported by: mrgabu
Review: https://reviewboard.asterisk.org/r/303/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204749 f38db490-d61c-443f-a65b-d21fe96a405b
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CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
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This merge splits the PRI/BRI signaling logic out of chan_dahdi.c into
sig_pri.c. Functionality in theory should not change (mostly). A few trivial
changes were made in sig_analog with verbose messages and commenting.
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