Merged revisions 337219 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/10

........
  r337219 | oej | 2011-09-21 11:32:50 +0200 (Ons, 21 Sep 2011) | 13 lines
  
  Make ast_pbx_run() not default to s@default if extension is not found
  
  Review: https://reviewboard.asterisk.org/r/1446/
  
  This is a bug - or architecture mistake - that has been in Asterisk for a 
  very long time. It was exposed by the AMI originate action and possibly
  some other applications. Most channel drivers checks if an extension
  exists BEFORE starting a pbx on an inbound call, so most calls will
  not depend on this issue.
  
  Thanks everyone involved in the review and on IRC and the mailing list
  for a quick review and all the feedback.

  (closes issue ASTERISK-18578)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337220 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/11.2
Olle Johansson 14 years ago
parent 2ae7ae00c8
commit 7b08b2cf53

@ -257,6 +257,21 @@ RTP Changes
* The rtp.conf setting "strictrtp" is now enabled by default. In previous
releases it was disabled.
PBX Core
--------
* The PBX core previously made a call with a non-existing extension test for
extension s@default and jump there if the extension existed.
This was a bad default behaviour and violated the principle of least surprise.
It has therefore been changed in this release. It may affect some
applications and configurations that rely on this behaviour. Most channel
drivers have avoided this for many releases by testing whether the extension
called exists before starting the PBX and generating a local error.
This behaviour still exists and works as before.
Extension "s" is used when no extension is given in a channel driver,
like immediate answer in DAHDI or calling to a domain with no user part
in a SIP uri.
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 1.6.2 to Asterisk 1.8 ----------------
------------------------------------------------------------------------------

@ -285,6 +285,10 @@ include => dundi-e164-switch
; In macros, it is the start extension. In most other cases,
; you have to goto "s" to execute that extension.
;
; Note: In old versions of Asterisk the PBX in some cases defaulted to
; extension "s" when a given extension was wrong (like in AMI originate).
; This is no longer the case.
;
; For wildcard matches, see above - all pattern matches start with
; an underscore.
exten => s,1,Goto(${ARG1},1)

@ -4977,23 +4977,16 @@ static enum ast_pbx_result __ast_pbx_run(struct ast_channel *c,
autoloopflag = ast_test_flag(c, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
ast_set_flag(c, AST_FLAG_IN_AUTOLOOP);
/* Start by trying whatever the channel is set to */
if (!ast_exists_extension(c, c->context, c->exten, c->priority,
S_COR(c->caller.id.number.valid, c->caller.id.number.str, NULL))) {
/* If not successful fall back to 's' */
if (ast_strlen_zero(c->exten)) {
/* If not successful fall back to 's' - but only if there is no given exten */
ast_verb(2, "Starting %s at %s,%s,%d failed so falling back to exten 's'\n", c->name, c->context, c->exten, c->priority);
/* XXX the original code used the existing priority in the call to
* ast_exists_extension(), and reset it to 1 afterwards.
* I believe the correct thing is to set it to 1 immediately.
*/
set_ext_pri(c, "s", 1);
if (!ast_exists_extension(c, c->context, c->exten, c->priority,
S_COR(c->caller.id.number.valid, c->caller.id.number.str, NULL))) {
/* JK02: And finally back to default if everything else failed */
ast_verb(2, "Starting %s at %s,%s,%d still failed so falling back to context 'default'\n", c->name, c->context, c->exten, c->priority);
ast_copy_string(c->context, "default", sizeof(c->context));
}
}
if (c->cdr) {
/* allow CDR variables that have been collected after channel was created to be visible during call */
ast_cdr_update(c);

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