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20 authors
  1. Jul 16, 2010
  3. Jul 15, 2010
  5. Jul 14, 2010
    • Jeff Peeler's avatar
      Fix MWI notification transmission problems over SIP. · 2b2a6123
      Jeff Peeler authored 
      MWI updates were not being sent if no messages were found in the event cache.
This was corrected since a phone may need to clear its MWI status configured
previously from another mailbox.

Upon module or sip reload, MWI updates could not be sent due to the sipsock
socket not being set early enough in reload_config. The code handling the
descriptor assignment and such has simply been moved before the call to
build_peer.

Issuing a sip reload cleared the IP address of the peer, but skipped checking
the database for registration information. The database is now checked both
for sip reload and actually reloading the module.

If a transmission occurs before the do_monitor thread has started, do not
attempt to send a signal to it.

(closes issue #17398)
Reported by: ip-rob


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2b2a6123
    • Mark Michelson's avatar
      Fix errors where incorrect address information was printed. · 1e8c66e7
      Mark Michelson authored 
      ast_sockaddr_stringiy_fmt (which is call by all ast_sockaddr_stringify* functions)
uses thread-local storage for storing the string that it creates. In cases where
ast_sockaddr_stringify_fmt was being called twice within the same statement, the
result of one call would be overwritten by the result of the other call. This
usually was happening in printf-like statements and was resulting in the same
stringified addressed being printed twice instead of two separate addresses.

I have fixed this by using ast_strdupa on the result of stringify functions if
they are used twice within the same statement. As far as I could tell, there were
no instances where a pointer to the result of such a call were saved anywhere, so
this is the only situation I could see where this error could occur.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276570 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      1e8c66e7
    • Richard Mudgett's avatar
      Make compile again. · 5d9aa457
      Richard Mudgett authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276531 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      5d9aa457
    • Tilghman Lesher's avatar
      Oops, merge reverted this fix. · 4c94d1ee
      Tilghman Lesher authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4c94d1ee
    • Tilghman Lesher's avatar
      Remove the old stub files, preferring the optional_api method. · 832d1296
      Tilghman Lesher authored 
      (closes issue #17475)
 Reported by: tilghman
 
Review: https://reviewboard.asterisk.org/r/695/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276490 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      832d1296
    • Kevin P. Fleming's avatar
      Don't try to call an embedded module's backup_globals() function until · 8e7d01d4
      Kevin P. Fleming authored 
      after confirming it exists.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276441 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8e7d01d4
    • David Vossel's avatar
      handle special case were "200 Ok" to pending INVITE never receives ACK · d1c9a4b3
      David Vossel authored 
      Unlike most responses, the 200 Ok to a pending INVITE Request is
acknowledged by an ACK Request.  If the ACK Request for this Response is not received
the previous behavior was to immediately destroy the dialog and hangup
the channel. Now in an effort to be more RFC compliant, instead of immediately
destroying the dialog during this special case, termination is done with a BYE Request
as the dialog is technically confirmed when the 200 Ok is sent even if the ACK is
never received.  The behavior of immediately hanging up the channel remains.
This only affects how dialog termination proceeds for this one special case.

RFC 3261 section 13.3.1.4
"If the server retransmits the 2xx response for 64*T1 seconds without receiving
an ACK, the dialog is confirmed, but the session SHOULD be terminated.  This is
accomplished with a BYE, as described in Section 15."



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      d1c9a4b3
    • Richard Mudgett's avatar
      Expand the caller ANI field to an ast_party_id · cf7bbcc4
      Richard Mudgett authored 
      Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.

This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/

Review: https://reviewboard.asterisk.org/r/744/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      cf7bbcc4
    • David Vossel's avatar
      collapse debug code in retrans_pkt into separate lines · e2599bc4
      David Vossel authored 
      I've been working in this function a bunch lately, and
these huge debug strings are getting annoying.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276392 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e2599bc4
    • Richard Mudgett's avatar
      Make compile again. · 2cf60bb0
      Richard Mudgett authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276391 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2cf60bb0
    • Jeff Peeler's avatar
      Do not skip sending MWI for a peer if an address is defined. Really just a merge mistake from IPv6 · f4c665ee
      Jeff Peeler authored 
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      f4c665ee
    • Tim Ringenbach's avatar
      Fix documentation for pgsql cel and cdr, and slightly improve pgsql_cel. · e19a6c24
      Tim Ringenbach authored 
      Change the documented pgsql schema to use "timestamp" instead of "time",
as the latter is only a time without a date.

Added some missing columns for cel's pgsql schema, and corrected spelling
on some others. Updated cel's uniqueid size to be the same as the cdr.
Added id column to cel's pgsql schema and updated code to allow unknown
columns to get their default value instead of forcing 0 or empty string.

Added microseconds to the timestamp cel logs to pgsql.

Review: https://reviewboard.asterisk.org/r/734


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e19a6c24
    • 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
    • Leif Madsen's avatar
      Merged revisions 276267 via svnmerge from · 608be652
      Leif Madsen authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r276267 | lmadsen | 2010-07-14 06:49:01 -0500 (Wed, 14 Jul 2010) | 1 line
  
  Update documentation for voicemail.conf externpass option.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      608be652
  7. Jul 13, 2010