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20 authors
  1. Sep 06, 2012
  3. Jul 16, 2012
  5. Feb 28, 2012
  7. Nov 14, 2011
  9. Nov 01, 2011
  11. Sep 19, 2011
    • Richard Mudgett's avatar
      Merged revisions 336659 via svnmerge from · 5c71a502
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r336659 | rmudgett | 2011-09-19 13:51:19 -0500 (Mon, 19 Sep 2011) | 38 lines
  
  Merged revisions 336658 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r336658 | rmudgett | 2011-09-19 13:46:40 -0500 (Mon, 19 Sep 2011) | 31 lines
    
    Made Dial d and H options no longer immediately auto-answer the calling leg.
    
    The Dial d and H options break DTMF attended transfer atxferdropcall
    option.
    
    1) Party A calls party B.
    2) Party B does a DTMF attended transfer to Party C.
    
    If the dialplan uses the Dial d or H options to call Party C then the Dial
    application answers the call immediately before initiating the call leg to
    Party C.  The premature answer causes the transfer code to not invoke the
    atxferdropcall=no behavior for a blonde transfer since Party C has
    "answered".  The transfer code thinks that Party B has "consulted" with
    Party C when Party B hangs up and completes the transfer to Party A.
    Party A now hears ringback until Party C actually answers.
    
    ASTERISK-13294 Dial d option.
    ASTERISK-11067 Dial H option to disconnect before answer.
    
    The referenced issues made Dial answer with the d and H options because
    many SIP and ISDN phones cannot send DTMF before the call is connected.
    
    * Made require the dialplan to control when or if the call needs to be
    answered to use the Dial application d and H options.  (The call is no
    longer surprise answered when using the Dial d or H options.)
    
    Review: https://reviewboard.asterisk.org/r/1381/
    
    JIRA AST-623
    JIRA AST-666
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@336662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      5c71a502
  13. Sep 16, 2011
  15. Jul 20, 2011
  17. May 25, 2011
    • Richard Mudgett's avatar
      Merged revisions 320823 via svnmerge from · 0096238b
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines
  
  The AMI Newstate event contains different information between v1.4 and v1.8.
  
  The addition of connected line support in v1.8 changes the behavior of the
  channel caller ID somewhat.  The channel caller ID value no longer time
  shares with the connected line ID on outgoing call legs.  The timing of
  some AMI events/responses output the connected line ID as caller ID.
  These party ID's are now separate.
  
  * The ConnectedLineNum and ConnectedLineName headers were added to many
  AMI events/responses if the CallerIDNum/CallerIDName headers were also
  present.
  
  (closes issue #18252)
  Reported by: gje
  Tested by: rmudgett
  
  Review: https://reviewboard.asterisk.org/r/1227/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0096238b
  19. Mar 04, 2011
    • Richard Mudgett's avatar
      Merged revisions 309445 via svnmerge from · 928ec2b9
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines
  
  Get real channel of a DAHDI call.
  
  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/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      928ec2b9
  21. Nov 10, 2010
  23. Sep 16, 2010
  25. Aug 24, 2010
  27. Aug 13, 2010
  29. Aug 10, 2010
  31. Jul 26, 2010
  33. Jul 23, 2010
  35. Jul 21, 2010
  37. Jul 14, 2010
    • Richard Mudgett's avatar
      ast_callerid restructuring · ec37ffbd
      Richard Mudgett authored 
      The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.

Eliminate struct ast_callerid and replace it with the following struct
organization:

struct ast_party_name {
	char *str;
	int char_set;
	int presentation;
	unsigned char valid;
};
struct ast_party_number {
	char *str;
	int plan;
	int presentation;
	unsigned char valid;
};
struct ast_party_subaddress {
	char *str;
	int type;
	unsigned char odd_even_indicator;
	unsigned char valid;
};
struct ast_party_id {
	struct ast_party_name name;
	struct ast_party_number number;
	struct ast_party_subaddress subaddress;
	char *tag;
};
struct ast_party_dialed {
	struct {
		char *str;
		int plan;
	} number;
	struct ast_party_subaddress subaddress;
	int transit_network_select;
};
struct ast_party_caller {
	struct ast_party_id id;
	char *ani;
	int ani2;
};

The new organization adds some new information as well.

* The party name and number now have their own presentation value that can
be manipulated independently.  ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.

* The party name and number now have a valid flag.  Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.

* The party name now has a character set value.  SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.

* The dialed party now has a numbering plan value that could be useful to
have available.

The various channel drivers will need to be updated to support the new
core features as needed.  They have simply been converted to supply
current functionality at this time.


The following items of note were either corrected or enhanced:

* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.

* CALLERPRES() is now deprecated because the name and number have their
own presentation values.

* Fixed app_alarmreceiver.c write_metadata().  The workstring[] could
contain garbage.  It also can only contain the caller id number so using
ast_callerid_parse() on it is silly.  There was also a typo in the
CALLERNAME if test.

* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string.  ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string.  Then using
ast_shrink_phone_number() could alter it even more.

* Fixed caller ID name and number memory leak in chan_usbradio.c.

* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.

* Protected access to a caller channel with lock in chan_sip.c.

* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk().  Also made save all caller ID data instead of just the name
and number strings.

* Simplified cdr.c set_one_cid().  It hand coded the ast_callerid_merge()
function.

* Corrected some weirdness with app_privacy.c's use of caller
presentation.

Review:	https://reviewboard.asterisk.org/r/702/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ec37ffbd
  39. Jul 06, 2010
  41. Jun 08, 2010
  43. Jun 03, 2010
  45. Jun 01, 2010
  47. May 28, 2010
  49. May 07, 2010
  51. Apr 21, 2010
  53. Feb 12, 2010
  55. Jan 18, 2010
  57. Dec 09, 2009
  59. Dec 02, 2009
  61. Nov 13, 2009
  63. Nov 02, 2009
    • Richard Mudgett's avatar
      DAHDI ISDN channel names will not allow device state to work. (Interim solution.) · 6406f395
      Richard Mudgett authored 
      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: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6406f395
  65. Oct 05, 2009
    • Kevin P. Fleming's avatar
      Allow non-compliant T.38 endpoints to be supportable via configuration option. · 20743ec0
      Kevin P. Fleming authored 
      Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.

This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.

In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).

(issue #15586)
Reported by: globalnetinc


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      20743ec0
  67. Oct 01, 2009
  69. Sep 24, 2009