Merge Call completion support into trunk.

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 An admittedly out-of-date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


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256528 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 52 insertions, 4 deletions

diff --git a/configs/chan_dahdi.conf.sample b/configs/chan_dahdi.conf.sample
index fb0d06931..30229b5fc 100644
--- a/configs/chan_dahdi.conf.sample
+++ b/configs/chan_dahdi.conf.sample
@@ -85,10 +85,11 @@
 ;service_message_support=yes
 ; Enable service message support for channel. Must be set after switchtype.
 ;
-; PRI Reverse Charging Indication: Indicate to the called party that the
-; call will be reverse charged.  To enable, prefix the dialed number with one
-; of the following letters:
-; C - Reverse Charge Indication Requested
+; Dialing options for ISDN (i.e., Dial(DAHDI/g1/exten/options)):
+; R      Reverse Charge Indication
+;          Indicate to the called party that the call will be reverse charged.
+; K(n)   Keypad digits n
+;          Send out the specified digits as keypad digits.
 ;
 ; PRI Dialplan: The ISDN-level Type Of Number (TON) or numbering plan, used for
 ; the dialed number.  For most installations, leaving this as 'unknown' (the
@@ -236,9 +237,52 @@
 ;       May vary in other ISDN standards (Q.931 1993 : 90000 ms)
 ; T313: Wait for CONNECT acknowledge, CPE side only (default 3000 ms)
 ;
+; T-RESPONSE:   Maximum time to wait for a typical APDU response. (default 4000 ms)
+;               This is an implementation timer when the standard does not specify one.
+; T-ACTIVATE:   Request supervision timeout. (default 10000 ms)
+; T-RETENTION:  Maximum  time to wait for user A to activate call-completion. (default 30000 ms)
+;               Used by ETSI PTP, ETSI PTMP, and Q.SIG as the cc_offer_timer.
+; T-CCBS1:      T-STATUS timer equivalent for CC user A status. (default 4000 ms)
+; T-CCBS2:      Maximum  time the CCBS service will be active (default 45 min in ms)
+; T-CCBS3:      Maximum  time to wait for user A to respond to user B availability. (default 20000 ms)
+; T-CCBS5:      Network B CCBS supervision timeout. (default 60 min in ms)
+; T-CCBS6:      Network A CCBS supervision timeout. (default 60 min in ms)
+; T-CCNR2:      Maximum  time the CCNR service will be active (default 180 min in ms)
+; T-CCNR5:      Network B CCNR supervision timeout. (default 195 min in ms)
+; T-CCNR6:      Network A CCNR supervision timeout. (default 195 min in ms)
+; CC-T1:        Q.SIG CC request supervision timeout. (default 30000 ms)
+; CCBS-T2:      Q.SIG CCBS supervision timeout. (default 60 min in ms)
+; CCNR-T2:      Q.SIG CCNR supervision timeout. (default 195 min in ms)
+; CC-T3:        Q.SIG CC Maximum time to wait for user A to respond to user B availability. (default 30000 ms)
+;
 ;pritimer => t200,1000
 ;pritimer => t313,4000
 ;
+; CC PTMP recall mode:
+; specific - Only the CC original party A can participate in the CC callback
+; global - Other compatible endpoints on the PTMP line can be party A in the CC callback
+;
+; cc_ptmp_recall_mode cannot be changed on a reload.
+;
+;cc_ptmp_recall_mode = specific
+;
+; CC Q.SIG Party A (requester) retain signaling link option
+; retain       Require that the signaling link be retained.
+; release      Request that the signaling link be released.
+; do_not_care  The responder is free to choose if the signaling link will be retained.
+;
+;cc_qsig_signaling_link_req = retain
+;
+; CC Q.SIG Party B (responder) retain signaling link option
+; retain       Prefer that the signaling link be retained.
+; release      Prefer that the signaling link be released.
+;
+;cc_qsig_signaling_link_rsp = retain
+;
+; See ccss.conf.sample for more options.  The timers described by ccss.conf.sample
+; are not used by ISDN for the native protocol since they are defined by the
+; standards and set by pritimer above.
+;
 ; To enable transmission of facility-based ISDN supplementary services (such
 ; as caller name from CPE over facility), enable this option.
 ; Cannot be changed on a reload.
@@ -267,6 +311,10 @@
 ; fxo_ks:         FXO (Kewl Start)
 ; pri_cpe:        PRI signalling, CPE side
 ; pri_net:        PRI signalling, Network side
+; bri_cpe:        BRI PTP signalling, CPE side
+; bri_net:        BRI PTP signalling, Network side
+; bri_cpe_ptmp:   BRI PTMP signalling, CPE side
+; bri_net_ptmp:   BRI PTMP signalling, Network side
 ; sf:             SF (Inband Tone) Signalling
 ; sf_w:           SF Wink
 ; sf_featd:       SF Feature Group D (The fake, Adtran style, DTMF)