From 63d49a667e802238e813f258b64618549d3293b4 Mon Sep 17 00:00:00 2001
From: Jim Dixon <telesistant@hotmail.com>
Date: Sun, 23 Mar 2003 17:14:29 +0000
Subject: [PATCH] Fixed so that dial from a Zap channel to a Zap channel in
'dataquality' mode actually puts channels into CLEAR mode (so that 56k ISDN
calls will work thru it) 64K calls STILL DONT.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
apps/app_dial.c | 6 +++++-
channels/chan_zap.c | 18 +++++++++++++++++-
include/asterisk/frame.h | 3 +++
3 files changed, 25 insertions(+), 2 deletions(-)
diff --git a/apps/app_dial.c b/apps/app_dial.c
index ac9953bc3e..e68f5137e7 100755
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -545,12 +545,16 @@ static int dial_exec(struct ast_channel *chan, void *data)
int x = 2;
if (tmp->dataquality) x = 0;
ast_channel_setoption(chan,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
+ x = 0;
+ ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
}
if (!strcmp(peer->type,"Zap"))
{
int x = 2;
if (tmp->dataquality) x = 0;
ast_channel_setoption(peer,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
+ x = 0;
+ ast_channel_setoption(chan,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
}
hanguptree(outgoing, peer);
outgoing = NULL;
@@ -573,7 +577,7 @@ static int dial_exec(struct ast_channel *chan, void *data)
ast_log(LOG_DEBUG, "app_dial: sendurl=%s.\n", url);
ast_channel_sendurl( peer, url );
} /* /JDG */
- res = ast_bridge_call(chan, peer, allowredir, allowdisconnect);
+ res = ast_bridge_call(chan, peer, allowredir, allowdisconnect | tmp->dataquality);
ast_hangup(peer);
}
out:
diff --git a/channels/chan_zap.c b/channels/chan_zap.c
index d762d87cde..a6a49f3bda 100755
--- a/channels/chan_zap.c
+++ b/channels/chan_zap.c
@@ -1617,6 +1617,8 @@ static int zt_hangup(struct ast_channel *ast)
x = 0;
ast_channel_setoption(ast,AST_OPTION_TONE_VERIFY,&x,sizeof(char),0);
ast_channel_setoption(ast,AST_OPTION_TDD,&x,sizeof(char),0);
+ x = 1;
+ ast_channel_setoption(ast,AST_OPTION_AUDIO_MODE,&x,sizeof(char),0);
p->didtdd = 0;
p->cidspill = NULL;
p->callwaitcas = 0;
@@ -1743,7 +1745,7 @@ int x;
struct zt_pvt *p = chan->pvt->pvt;
- if ((option != AST_OPTION_TONE_VERIFY) &&
+ if ((option != AST_OPTION_TONE_VERIFY) && (option != AST_OPTION_AUDIO_MODE) &&
(option != AST_OPTION_TDD) && (option != AST_OPTION_RELAXDTMF))
{
errno = ENOSYS;
@@ -1857,6 +1859,20 @@ int x;
}
ast_dsp_digitmode(p->dsp,x ? DSP_DIGITMODE_RELAXDTMF : DSP_DIGITMODE_DTMF | p->dtmfrelax);
break;
+ case AST_OPTION_AUDIO_MODE: /* Set AUDIO mode (or not) */
+ if (!*cp)
+ {
+ ast_log(LOG_DEBUG, "Set option AUDIO MODE, value: OFF(0) on %s\n",chan->name);
+ x = 0;
+ }
+ else
+ {
+ ast_log(LOG_DEBUG, "Set option AUDIO MODE, value: ON(1) on %s\n",chan->name);
+ x = 1;
+ }
+ if (ioctl(p->subs[SUB_REAL].zfd, ZT_AUDIOMODE, &x) == -1)
+ ast_log(LOG_WARNING, "Unable to set audio mode on channel %d\n", p->channel);
+ break;
}
errno = 0;
return 0;
diff --git a/include/asterisk/frame.h b/include/asterisk/frame.h
index e971a6d034..2dbcccf086 100755
--- a/include/asterisk/frame.h
+++ b/include/asterisk/frame.h
@@ -188,6 +188,9 @@ struct ast_frame_chain {
/* Relax the parameters for DTMF reception (mainly for radio use) */
#define AST_OPTION_RELAXDTMF 3
+/* Set (or clear) Audio (Not-Clear) Mode */
+#define AST_OPTION_AUDIO_MODE 4
+
struct ast_option_header {
/* Always keep in network byte order */
#if __BYTE_ORDER == __BIG_ENDIAN
--
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