From: Richard Mudgett <rmudgett@digium.com>
Date: Thu, 9 Jul 2009 23:37:53 +0000 (+0000)
Subject: No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
X-Git-Tag: 1.4.26-rc6~8
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=fb1c512a401d5902c84a01c20f675ba7854e1e62;p=thirdparty%2Fasterisk.git

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/1.4@205728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_dahdi.c b/channels/chan_dahdi.c
index d10ddb905d..70830e0248 100644
--- a/channels/chan_dahdi.c
+++ b/channels/chan_dahdi.c
@@ -9670,6 +9670,7 @@ static void *pri_dchannel(void *vpri)
 						} else if (pri->pvts[chanpos]->confirmanswer) {
 							ast_log(LOG_DEBUG, "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]);