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20 authors
  1. Apr 12, 2010
  3. Apr 11, 2010
  5. Apr 10, 2010
  7. Apr 09, 2010
    • Mark Michelson's avatar
      Remove status_response callbacks where they are not needed. · 9afa6af8
      Mark Michelson authored 
      
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      9afa6af8
    • Mark Michelson's avatar
      Prevent crash when originating a call to a local channel. · 75099496
      Mark Michelson authored 
      Call completion code tries to grab the call completion parameters
from the requesting channel during local_request. When originating
a call to a local channel, however, this channel is NULL. This
was causing an issue for me when trying to run a test script.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      75099496
    • Richard Mudgett's avatar
      Merge CCSS architecture document from CCSS branch. · d66b44b4
      Richard Mudgett authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256608 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d66b44b4
    • Richard Mudgett's avatar
      Remove PRI CCSS BUGBUG message and update configure script. · 537edff1
      Richard Mudgett authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      537edff1
    • Mark Michelson's avatar
      Add routines for parsing SIP URIs consistently. · 4b8f1c8c
      Mark Michelson authored 
      From the original issue report opened by Nick Lewis:
Many sip headers in many sip methods contain the ABNF structure
 name-andor-addr = name-addr / addr-spec
 Examples include the to-header, from-header, contact-header, replyto-header

 At the moment chan_sip.c makes various different attempts to parse this name-andor-addr structure for each header type and for each sip method with sometimes limited degrees of success.

 I recommend that this name-andor-addr structure be parsed by a dedicated function and that it be used irrespective of the specific method or header that contains the name-andor-addr structure

Nick has also included unit tests for verifying these routines as well, so...heck yeah.

(closes issue #16708)
Reported by: Nick_Lewis
Patches:
      reqresp_parser-nameandoraddr2.patch uploaded by Nick Lewis (license 657

Review: https://reviewboard.asterisk.org/r/549



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4b8f1c8c
    • Mark Michelson's avatar
      Fix some compiler errors that popped up after the CCSS merge. · ae7b76a1
      Mark Michelson authored 
      
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ae7b76a1
    • Mark Michelson's avatar
      Merge Call completion support into trunk. · e24661fd
      Mark Michelson authored 
      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: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e24661fd
    • Mark Michelson's avatar
      func_srv and explicit specification of a remote IP for SIP. · 6cad0f16
      Mark Michelson authored 
      From Review Board:
There are two interrelated changes here.

First, there is the introduction of func_srv. This adds two new read-only
dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the
ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV
records instead. In order to facilitate this work, I added a couple of new API
calls to srv.h. ast_srv_get_record_count tells the number of records returned
by an SRV lookup. This number is calculated at the time of the SRV lookup.
ast_srv_get_nth_record allows one to get a numbered SRV record.

Second, there is the modification to chan_sip that allows one to specify a
hostname or IP address (along with a port) to send an outgoing INVITE to when
dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV
records and then use the host and port from the results to dial via a specific
host instead of what is configured in sip.conf.

Review: https://reviewboard.asterisk.org/r/608
SWP-1200



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6cad0f16
  9. Apr 08, 2010
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  21. Mar 31, 2010
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  25. Mar 29, 2010
  27. Mar 27, 2010
  29. Mar 26, 2010