From b81f233e68e71d4601b932eb8b51b32b4312f231 Mon Sep 17 00:00:00 2001
From: Matthew Fredrickson <creslin@digium.com>
Date: Fri, 1 Apr 2005 17:00:50 +0000
Subject: [PATCH] Merging in xylome's beaerer capabilty patch (bug 3547)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@5342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
apps/Makefile | 2 +-
apps/app_dial.c | 8 +++---
channel.c | 20 +++++++++++++++
channels/chan_zap.c | 49 +++++++++----------------------------
include/asterisk/channel.h | 15 +++++++++++-
include/asterisk/transcap.h | 33 +++++++++++++++++++++++++
6 files changed, 84 insertions(+), 43 deletions(-)
create mode 100755 include/asterisk/transcap.h
diff --git a/apps/Makefile b/apps/Makefile
index 0feec316d5..2f6a875dcb 100755
--- a/apps/Makefile
+++ b/apps/Makefile
@@ -31,7 +31,7 @@ APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_mp3.so\
app_talkdetect.so app_alarmreceiver.so app_userevent.so app_verbose.so \
app_test.so app_forkcdr.so app_math.so app_realtime.so \
app_dumpchan.so app_waitforsilence.so app_while.so app_setrdnis.so \
- app_md5.so app_readfile.so app_chanspy.so
+ app_md5.so app_readfile.so app_chanspy.so app_settransfercapability.so
ifneq (${OSARCH},Darwin)
ifneq (${OSARCH},SunOS)
diff --git a/apps/app_dial.c b/apps/app_dial.c
index 9ee512ca55..9f7dedbdda 100755
--- a/apps/app_dial.c
+++ b/apps/app_dial.c
@@ -77,8 +77,8 @@ static char *descrip =
" 'r' -- indicate ringing to the calling party, pass no audio until answered.\n"
" 'm[(class)]' -- provide hold music to the calling party until answered (optionally\n"
" with the specified class.\n"
-" 'M(x[^arg]) -- Executes the macro (x with ^ delim arg list) upon connect of the call.\n"
-" Also, the macro can set the MACRO_RESULT variable to do the following:\n"
+" 'M(x[^arg])' -- Executes the macro (x with ^ delim arg list) upon connect of the call.\n"
+" Also, the macro can set the MACRO_RESULT variable to do the following:\n"
" -- ABORT - Hangup both legs of the call.\n"
" -- CONGESTION - Behave as if line congestion was encountered.\n"
" -- BUSY - Behave as if a busy signal was encountered. (n+101)\n"
@@ -1039,8 +1039,8 @@ static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags
tmp->chan->cid.cid_tns = chan->cid.cid_tns;
/* Presense of ADSI CPE on outgoing channel follows ours */
tmp->chan->adsicpe = chan->adsicpe;
- /* pass the digital flag */
- ast_copy_flags(tmp->chan, chan, AST_FLAG_DIGITAL);
+ /* Pass the transfer capability */
+ tmp->chan->transfercapability = chan->transfercapability;
/* If we have an outbound group, set this peer channel to it */
if (outbound_group)
diff --git a/channel.c b/channel.c
index e621aec5e1..460d1b7814 100755
--- a/channel.c
+++ b/channel.c
@@ -39,6 +39,7 @@
#include <asterisk/utils.h>
#include <asterisk/lock.h>
#include <asterisk/app.h>
+#include <asterisk/transcap.h>
#ifdef ZAPTEL_OPTIMIZATIONS
#include <sys/ioctl.h>
#ifdef __linux__
@@ -243,6 +244,25 @@ char *ast_state2str(int state)
}
}
+char *ast_transfercapability2str(int transfercapability)
+{
+ switch(transfercapability) {
+ case AST_TRANS_CAP_SPEECH:
+ return "SPEECH";
+ case AST_TRANS_CAP_DIGITAL:
+ return "DIGITAL";
+ case AST_TRANS_CAP_RESTRICTED_DIGITAL:
+ return "RESTRICTED_DIGITAL";
+ case AST_TRANS_CAP_3_1K_AUDIO:
+ return "3K1AUDIO";
+ case AST_TRANS_CAP_DIGITAL_W_TONES:
+ return "DIGITAL_W_TONES";
+ case AST_TRANS_CAP_VIDEO:
+ return "VIDEO";
+ default:
+ return "UNKNOWN";
+ }
+}
int ast_best_codec(int fmts)
{
diff --git a/channels/chan_zap.c b/channels/chan_zap.c
index c68f0ff488..8eb891141b 100755
--- a/channels/chan_zap.c
+++ b/channels/chan_zap.c
@@ -38,6 +38,7 @@
#include <asterisk/causes.h>
#include <asterisk/term.h>
#include <asterisk/utils.h>
+#include <asterisk/transcap.h>
#include <sys/signal.h>
#include <errno.h>
#include <stdlib.h>
@@ -1866,13 +1867,15 @@ static int zt_call(struct ast_channel *ast, char *rdest, int timeout)
ast_log(LOG_DEBUG, "I'm being setup with no bearer right now...\n");
pri_set_crv(p->pri->pri, p->call, p->channel, 0);
}
- p->digital = ast_test_flag(ast,AST_FLAG_DIGITAL);
+ p->digital = IS_DIGITAL(ast->transfercapability);
pri_sr_set_channel(sr, p->bearer ? PVT_TO_CHANNEL(p->bearer) : PVT_TO_CHANNEL(p),
p->pri->nodetype == PRI_NETWORK ? 0 : 1, 1);
- pri_sr_set_bearer(sr, p->digital ? PRI_TRANS_CAP_DIGITAL : PRI_TRANS_CAP_SPEECH,
+ pri_sr_set_bearer(sr, p->digital ? PRI_TRANS_CAP_DIGITAL : ast->transfercapability,
(p->digital ? -1 :
((p->law == ZT_LAW_ALAW) ? PRI_LAYER_1_ALAW : PRI_LAYER_1_ULAW)));
- dp_strip = 0;
+ if (option_verbose > 2)
+ ast_verbose(VERBOSE_PREFIX_3 "Requested transfer capability: 0x%.2x - %s\n", ast->transfercapability, ast_transfercapability2str(ast->transfercapability));
+ dp_strip = 0;
pridialplan = p->pri->dialplan - 1;
if (pridialplan == -2) { /* compute dynamically */
if (strncmp(c + p->stripmsd, p->pri->internationalprefix, strlen(p->pri->internationalprefix)) == 0) {
@@ -4604,35 +4607,7 @@ static int zt_indicate(struct ast_channel *chan, int condition)
return res;
}
-#ifdef ZAPATA_PRI
-static void set_calltype(struct ast_channel *chan, int ctype)
-{
- char *s = "UNKNOWN";
- switch(ctype) {
- case PRI_TRANS_CAP_SPEECH:
- s = "SPEECH";
- break;
- case PRI_TRANS_CAP_DIGITAL:
- s = "DIGITAL";
- break;
- case PRI_TRANS_CAP_RESTRICTED_DIGITAL:
- s = "RESTRICTED_DIGITAL";
- break;
- case PRI_TRANS_CAP_3_1K_AUDIO:
- s = "31KAUDIO";
- break;
- case PRI_TRANS_CAP_7K_AUDIO:
- s = "7KAUDIO";
- break;
- case PRI_TRANS_CAP_VIDEO:
- s = "VIDEO";
- break;
- }
- pbx_builtin_setvar_helper(chan, "CALLTYPE", s);
-}
-#endif
-
-static struct ast_channel *zt_new(struct zt_pvt *i, int state, int startpbx, int index, int law, int ctype)
+static struct ast_channel *zt_new(struct zt_pvt *i, int state, int startpbx, int index, int law, int transfercapability)
{
struct ast_channel *tmp;
int deflaw;
@@ -4764,14 +4739,14 @@ static struct ast_channel *zt_new(struct zt_pvt *i, int state, int startpbx, int
tmp->cid.cid_pres = i->callingpres;
tmp->cid.cid_ton = i->cid_ton;
#ifdef ZAPATA_PRI
- set_calltype(tmp, ctype);
+ tmp->transfercapability = transfercapability;
+ pbx_builtin_setvar_helper(tmp, "TRANSFERCAPABILITY", ast_transfercapability2str(transfercapability));
+ if (transfercapability & PRI_TRANS_CAP_DIGITAL) {
+ i->digital = 1;
+ }
/* Assume calls are not idle calls unless we're told differently */
i->isidlecall = 0;
i->alreadyhungup = 0;
- if (ctype & PRI_TRANS_CAP_DIGITAL) {
- i->digital = 1;
- ast_set_flag(tmp, AST_FLAG_DIGITAL);
- }
#endif
/* clear the fake event in case we posted one before we had ast_chanenl */
i->fake_event = 0;
diff --git a/include/asterisk/channel.h b/include/asterisk/channel.h
index 92c5000313..4d121bb69e 100755
--- a/include/asterisk/channel.h
+++ b/include/asterisk/channel.h
@@ -315,6 +315,9 @@ struct ast_channel {
/*! channel flags of AST_FLAG_ type */
unsigned int flags;
+ /* ISDN Transfer Capbility - AST_FLAG_DIGITAL is not enough */
+ unsigned short transfercapability;
+
struct ast_frame *readq;
int alertpipe[2];
/*! Write translation path */
@@ -338,7 +341,8 @@ struct ast_channel {
/* Channels have this property if they can accept input with jitter; i.e. most VoIP channels */
#define AST_CHAN_TP_WANTSJITTER (1 << 0)
-#define AST_FLAG_DIGITAL (1 << 0) /* if the call is a digital ISDN call */
+/* This flag has been deprecated by the transfercapbilty data member in struct ast_channel */
+/* #define AST_FLAG_DIGITAL (1 << 0) */ /* if the call is a digital ISDN call */
#define AST_FLAG_DEFER_DTMF (1 << 1) /* if dtmf should be deferred */
#define AST_FLAG_WRITE_INT (1 << 2) /* if write should be interrupt generator */
#define AST_FLAG_BLOCKING (1 << 3) /* if we are blocking */
@@ -811,6 +815,15 @@ int ast_channel_masquerade(struct ast_channel *original, struct ast_channel *clo
*/
char *ast_state2str(int state);
+/*! Gives the string form of a given transfer capability */
+/*!
+ * \param transercapability transfercapabilty to get the name of
+ * Give a name to a transfercapbility
+ * See above
+ * Returns the text form of the binary transfer capbility
+ */
+char *ast_transfercapability2str(int transfercapability);
+
/* Options: Some low-level drivers may implement "options" allowing fine tuning of the
low level channel. See frame.h for options. Note that many channel drivers may support
none or a subset of those features, and you should not count on this if you want your
diff --git a/include/asterisk/transcap.h b/include/asterisk/transcap.h
new file mode 100755
index 0000000000..1a77dd1ec8
--- /dev/null
+++ b/include/asterisk/transcap.h
@@ -0,0 +1,33 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * General Asterisk channel definitions.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Matthew Fredrickson <creslin@digium.com>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#ifndef _ASTERISK_TRANSCAP_H
+#define _ASTERISK_TRANSCAP_H
+
+/* These definitions are taken directly out of libpri.h and used here.
+ * DO NOT change them as it will cause unexpected behavior in channels
+ * that utilize these fields.
+ */
+
+#define AST_TRANS_CAP_SPEECH 0x0
+#define AST_TRANS_CAP_DIGITAL 0x08
+#define AST_TRANS_CAP_RESTRICTED_DIGITAL 0x09
+#define AST_TRANS_CAP_3_1K_AUDIO 0x10
+#define AST_TRANS_CAP_7K_AUDIO 0x11 /* Depriciated ITU Q.931 (05/1998)*/
+#define AST_TRANS_CAP_DIGITAL_W_TONES 0x11
+#define AST_TRANS_CAP_VIDEO 0x18
+
+#define IS_DIGITAL(cap)\
+ (cap) & AST_TRANS_CAP_DIGITAL ? 1 : 0
+
+#endif /* _ASTERISK_TRANSCAP_H */
--
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