]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Merged revisions 205728 via svn merge from
authorRichard Mudgett <rmudgett@digium.com>
Thu, 9 Jul 2009 23:51:50 +0000 (23:51 +0000)
committerRichard Mudgett <rmudgett@digium.com>
Thu, 9 Jul 2009 23:51:50 +0000 (23:51 +0000)
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r205728 | rmudgett | 2009-07-09 18:37:53 -0500 (Thu, 09 Jul 2009) | 21 lines

  No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.

  Add missing clearing of the dialing flag when the ISDN call is CONNECTED.
  (i.e. When libpri generates the event PRI_EVENT_ANSWER.)

  (closes issue #15420)
  Reported by: scottbmilne
  Patches:
        bug15420-1.4.25.1-diff2.txt uploaded by alecdavis (license 585)
  Tested by: scottbmilne, alecdavis

  (closes issue #15416)
  Reported by: avinoash

  (closes issue #15389)
  Reported by: alecdavis

  This patch should also fix the following issue:
  (issue #15205)
  Reported by: vinsik
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@205730 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_dahdi.c

index bc2ff640c03e0fac182a8dd0146c642bb30c3eb6..1c596887c533cd6e5b429ac3760faa2f42d5597d 100644 (file)
@@ -11681,6 +11681,7 @@ static void *pri_dchannel(void *vpri)
                                                } else if (pri->pvts[chanpos]->confirmanswer) {
                                                        ast_debug(1, "Waiting on answer confirmation on channel %d!\n", pri->pvts[chanpos]->channel);
                                                } else {
+                                                       pri->pvts[chanpos]->dialing = 0;
                                                        pri->pvts[chanpos]->subs[SUB_REAL].needanswer =1;
                                                        /* Enable echo cancellation if it's not on already */
                                                        dahdi_enable_ec(pri->pvts[chanpos]);